| 00:00.14 | JacksLivr | Strom_C: that hurts |
| 00:01.09 | DrukenLPY | anyone get screwed over the timechange today ? |
| 00:01.36 | JacksLivr | i got screwed out of 1 hour of sweet slumber |
| 00:02.24 | DrukenLPY | i didn't :) |
| 00:02.40 | DrukenLPY | i was fast asleep long before i normally am.. hehehe |
| 00:02.48 | *** part/#asterisk Strom_C (n=strom@66.0.239.106) |
| 00:03.16 | JacksLivr | my wife read a book till 1am and the light kept me up |
| 00:03.49 | JacksLivr | anyone see anything terrible in my dhcp config? |
| 00:04.11 | JacksLivr | 7910 stuck on configuring ip |
| 00:05.11 | *** join/#asterisk Strom_M (n=strom@66.0.239.106) |
| 00:05.18 | JacksLivr | ive about spent all the time i'm willing to on this phone |
| 00:05.59 | JacksLivr | if my time is worth 3.99/hr, i just about could bye a new phone |
| 00:06.15 | *** join/#asterisk stridernzl (n=neville@222-155-187-49.jetstream.xtra.co.nz) |
| 00:12.25 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:16.53 | digime | who is everyone using these days for toll free DID? |
| 00:17.29 | jql | I'll take two. :) |
| 00:18.10 | kink0 | I have to go, will tell you how D channel issue ends. Thanks for your help. |
| 00:19.06 | sudhir492 | Does anyone have firmware for Cisco 7940 ? |
| 00:21.31 | *** join/#asterisk saftsack (n=oliver@pD9E04E66.dip.t-dialin.net) |
| 00:28.57 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:28.58 | *** join/#asterisk coppice (n=chatzill@74.197.17.210.dyn.pacific.net.hk) |
| 00:37.21 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 00:46.09 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 00:54.31 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 00:59.43 | Hmmhesays | anyone in here use a sangoma a101? |
| 01:02.21 | ltdwk | i doubt it, however sangoma have great support. email or call them! |
| 01:02.36 | Hmmhesays | yeah I'm thinking about going with sangoma for my next t1 card |
| 01:02.46 | ltdwk | i use their multiport e1 card |
| 01:03.07 | ltdwk | but for Frame Relay |
| 01:03.16 | ltdwk | it just keeps going and going... stable as a rock |
| 01:04.04 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:04.23 | JT | multiport e1? i thought all their cards were t1 and e1 capable, or is it an older card? |
| 01:04.54 | ltdwk | it does both |
| 01:05.06 | JT | right |
| 01:05.37 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 01:06.01 | ltdwk | it's the S514-7 |
| 01:06.14 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 01:08.23 | ltdwk | Sangoma were quite willing to sell to us at dealer rates too. I think that card cost around $600 USD at dealer rates |
| 01:09.51 | ezway | sangoma are really impressive ! |
| 01:10.11 | ltdwk | and they stand by their products... |
| 01:10.23 | JunK-Y | hey Hmmhesays ! |
| 01:10.37 | JunK-Y | whats up dude? |
| 01:14.29 | Hmmhesays | not much JunK-Y: how are you? |
| 01:16.13 | JunK-Y | im fine, just came back to sherbrooke now. |
| 01:16.29 | Hmmhesays | how is that going for you |
| 01:17.04 | JunK-Y | going great, 1.5 months to go before huge homework are back. |
| 01:26.49 | JT | 184-240 calls dropped if comething goes wrong on the machine... yay! |
| 01:27.08 | JT | i mean, the density is pretty cool |
| 01:27.16 | JT | but seems like too many eggs in one basket |
| 01:27.38 | ltdwk | agreed...would have to be a very powerful machine to handle that many call without glitches |
| 01:29.25 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:37.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
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| 01:44.33 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 01:45.01 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 01:46.57 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 01:51.10 | *** join/#asterisk mceGEEK (n=nospam@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 01:51.21 | mceGEEK | hello |
| 01:52.13 | mceGEEK | how do i use queues .. i setup 1 queue in asterisknow with 2 members in it |
| 01:52.41 | *** join/#asterisk f40 (i=f40@adsl-76-214-194-50.dsl.emhril.sbcglobal.net) |
| 01:53.14 | sudhir492 | Can someone email the files to program Cisco 7940 phones? |
| 01:53.24 | Sweeper | JT: it's more for redundancy |
| 01:53.27 | Sweeper | extra ports |
| 01:53.44 | f40 | hi, anybody from digium in here? |
| 01:54.09 | JT | Sweeper: well it's only really redundant if you have two machines and an L1 failover box |
| 01:54.37 | Sweeper | different sort of redundancy |
| 01:55.04 | sudhir492 | f40: I am not from digium, but what do you need? |
| 01:55.25 | Sweeper | we'll use 4 ports, and have 4 extra ports that can be used if a port fails without replacing the card |
| 01:55.36 | Sweeper | of course, if a port fails, the card will be replaced eventually |
| 01:55.40 | JT | Sweeper: port on the card? |
| 01:55.44 | Sweeper | yea |
| 01:55.59 | Sweeper | t1 ports fail *shrug* |
| 01:56.07 | f40 | sudhir492: i recently switched * boxes and need them to reset my g729 licenses |
| 01:56.07 | JacksLivr | 0h..... im getting closer on the 7910 |
| 01:56.08 | mceGEEK | anyone using asterisknow? |
| 01:56.25 | *** join/#asterisk lba (n=lba@user-0ccsqlo.cable.mindspring.com) |
| 01:56.28 | Sweeper | mceGEEK: if you want support, different channel |
| 01:56.32 | JT | Sweeper: how often have you had this fail? |
| 01:56.44 | f40 | Sudhir492: also i have config files for cisco 7960G for sip |
| 01:56.45 | Sweeper | JT: personally, only had a single port go bad once |
| 01:56.57 | Sweeper | but I've only been dealing with this stuff for 10 months |
| 01:57.20 | JT | seems like a rare occurence |
| 01:57.24 | Sweeper | eh |
| 01:57.55 | JT | you're the first person i've heard who's had a port on a pri card go bad |
| 01:58.05 | Sweeper | my superiors say "buy it", I just give the model numbers to the inventory people :P |
| 01:58.24 | JT | going by that logic it's better to buy 2 X 4 port cards than 1 X 8 port card |
| 01:58.30 | mceGEEK | Sweeper: no one in asterisknow at this moment |
| 01:58.31 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 01:58.31 | Sweeper | true! |
| 01:58.56 | mceGEEK | Sweeper: done queues? |
| 01:59.03 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 02:00.34 | Sweeper | mceGEEK: in real asterisk, sure |
| 02:00.52 | Sweeper | they're really quite easy |
| 02:01.42 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 02:01.54 | Sweeper | just read the comments in queues.conf and agents.conf |
| 02:02.12 | ltdwk | I had probs with queues... queues would get stuck and calls would pile up |
| 02:02.19 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 02:02.20 | Sweeper | http://www.orderlyq.com/asteriskqueues.html <-- |
| 02:02.24 | Sweeper | and read that |
| 02:02.39 | Sweeper | ltdwk: really? |
| 02:02.41 | quidpro | Hmm... is there a way for * to know if an analog extension is already in use on a channel before dialing out? (ie. mixed analog and * on one phone line)? |
| 02:02.44 | Sweeper | what version of asterisk? |
| 02:02.50 | ltdwk | 1.2.10 |
| 02:02.54 | Sweeper | mmm |
| 02:02.59 | Sweeper | interesting |
| 02:03.05 | Sweeper | I'll keep a close eye on ours |
| 02:03.17 | sudhir492 | f40: If it is the first time you are doing, you dont have to call them. |
| 02:03.18 | *** part/#asterisk f40 (i=f40@adsl-76-214-194-50.dsl.emhril.sbcglobal.net) |
| 02:03.18 | Sweeper | we're in demo phase atm, so not much volum |
| 02:03.32 | ltdwk | I was putting a high volume through it and it was choking |
| 02:03.47 | Sweeper | define high? |
| 02:04.05 | JT | quidpro: no |
| 02:04.06 | ltdwk | well, not realy that high... probably around 10 to 20 calls per minute |
| 02:04.14 | JT | quidpro: asterisk assumes it's the only thing on a line |
| 02:05.30 | Sweeper | JT: isn't there a way to listen for a bit, and check to see if it's dialtone? |
| 02:05.33 | sudhir492 | JT: Depending on the density you need, octal card is desirable sometimes |
| 02:05.35 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 02:05.47 | quidpro | JT: Hehe... I Thought so. Walking around with the blinders on. I wish it could be the only thing on the line, but I am not rich enough to conver the whole house over to SIP phones. |
| 02:06.01 | JT | sudhir492: you'd want redundancy though |
| 02:06.10 | JT | Sweeper: not that i'm aware of |
| 02:06.10 | Sweeper | I guess you could do it ghetto-stlye, and use a wav-file comparitor program, record dialtone, and then run a record for a few seconds and compare |
| 02:06.20 | Sweeper | and then dial if :D |
| 02:06.24 | JT | quidpro: well if it's just voice, you could make the rest of the house run on one FXS port |
| 02:06.30 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
| 02:06.32 | sudhir492 | Even in quad case, you have the possibility of 100 calls dropping at once |
| 02:06.52 | quidpro | Sweeper: I remember my Commodore 64 1620 (I think that as the model #) 1200 baud modem could detect dialtone before tialing out. |
| 02:06.54 | Sweeper | 92, but whatever >.> |
| 02:07.00 | Sweeper | quidpro: hey! |
| 02:07.04 | Sweeper | there ya go |
| 02:07.06 | Sweeper | plug in a modem |
| 02:07.09 | Sweeper | write a script |
| 02:07.11 | quidpro | JT: Good idea, never thought of it that way |
| 02:07.16 | sudhir492 | At present I have 8 PRIs and I use three boxes with quad cards, two in production and 1 as backup |
| 02:07.29 | JacksLivr | anyone have a macaddr.cnf file for a skinny phone that they'd share? |
| 02:07.43 | JacksLivr | i think getting this right will be all i need to get the 7910 working |
| 02:07.59 | sudhir492 | BUt someone with 4 times bigger than me would prefer to have Octal card |
| 02:08.21 | JT | sudhir492: since nothing can do pri in-progress call failover, i think 8 pris is too dangerous |
| 02:09.04 | JT | sudhir492: it makes sense to get lots of compact embedded-like system running lower density (up to 4 port) cards acting as media gateways |
| 02:09.59 | sudhir492 | I agree with you. There is another factor: for performance reasons, you will need to have echo suppression in hardware, and if you are doing transcoding, octal becomes too demanding even with hardware silence suppression. |
| 02:10.39 | JT | do you mean echo supression not silence supression? |
| 02:10.51 | sudhir492 | Instead, you can do silence suppression in software and transcoding, still have enough muscle in dual core dual xeon/opteron box under $2000 |
| 02:11.17 | sudhir492 | so pricewise, quad cards win! |
| 02:12.07 | JT | i don't think you need anything near that much power if you're just media gatewaying to g.711 |
| 02:12.28 | Sweeper | OH |
| 02:12.47 | Sweeper | we're running a lot of satellite services |
| 02:13.18 | Sweeper | depending on the customer, they might need only a few channels off of some t1 hardware |
| 02:13.29 | Sweeper | or maybe we have a partial t1 coming in from elsewhere |
| 02:13.38 | Sweeper | so we need ports, but channels won't be used |
| 02:13.39 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 02:14.34 | sudhir492 | JT: Yes, echo suppression |
| 02:14.43 | Sweeper | while I'd prefer to give them all SIP phones, can't always do so |
| 02:15.38 | JT | sudhir492: should be done in hardware imho |
| 02:16.18 | JT | Sweeper: how does the sat service work? |
| 02:16.45 | Sweeper | JT: there's no single form it runs in |
| 02:17.19 | Sweeper | we've got SCPC links that take v.35, consumer platforms like idirect and gilat, and hybrids like vipersat |
| 02:17.19 | sudhir492 | JT: Why? |
| 02:17.36 | Sweeper | we just acquired a company who's motto was "give us a PO, and we'll do it" |
| 02:17.50 | Sweeper | so the support issues just tripled |
| 02:17.53 | sudhir492 | software echo cancellation works fine. Moreover, if you are in US, most of the Telco do echo cancellation anyway. |
| 02:18.07 | Sweeper | a popular platform is RAD's VMUX's |
| 02:18.10 | Zand3r | Hi all - I have a couple of Polycom 501 SIP phones and some Polycom 430 SIP phones. I have a Digium TDM400P leading out to two analogue lines and I have a single voicemail box configured which all phones will share. |
| 02:18.10 | Zand3r | The correct phones ring when an incoming call is placed and voicemail can be left when appropriate. I can also dial extension 200 from any phone to check voicemail. |
| 02:18.10 | Zand3r | I have however run in to a brick wall trying to have the phones notify me when a new voicemail has been left. I added "type=friend" and "mailbox=200" for each phone under Asterisk's sip.conf and for the phones display the message details correctly. When I use these two configuration lines however, the Polycom 430 phones completely freeze and the Polycom 501 phones can dial internal extensions but not outside numbers (I dia |
| 02:18.12 | Zand3r | l 9 to signify a call to the outside). |
| 02:18.15 | JT | sudhir492: hardware ec is more effective than software ec |
| 02:18.28 | JT | Zand3r: did you make that in a text editor first? |
| 02:18.33 | Sweeper | which are essentially 2-wire-over-ip, with t1 options |
| 02:18.57 | JT | Sweeper: nie |
| 02:19.28 | Zand3r | JT - no, I just installed XCHAT (windows) which I have not used for many years and for some reason I just noticed it did not submit the lines properly and when I hit return the final time it looks like they all flew down together. Sorry, I wasn;t intendign to flood - I just look at my keyboard when I type so didn;t notice. |
| 02:19.55 | JT | Zand3r: ah ok :) |
| 02:22.05 | flenders | saw you guys talking about echo cancellation on software vs. hardware.... I spent my whole sunday messing around with rx/tx gains, to get rid of some echo, and it's not 100% gone yet |
| 02:22.19 | flenders | I'll never get a card without EC again |
| 02:22.31 | Sweeper | Zand3r: did you also put the mailbox info in the polycom configs? |
| 02:22.40 | sudhir492 | Hmmhesays: what kind of card are you using? |
| 02:22.43 | JT | so true, the people who say you don't need hardware EC have just been lucky so far |
| 02:23.08 | Sweeper | Zand3r: http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk |
| 02:23.35 | flenders | now, with echotraining and echocancell, echo goes away after 2 seconds on the call, and even before that, it's not as bad as it was... |
| 02:23.36 | [TK]D-Fender | Zand3r: do "mailbox=number@context" |
| 02:24.00 | Zand3r | Sweeper: I set msg.mwi.1.callBack="200" which allowed me to use the "Messages" button to check voicemail from any phone but I did not see anywhere else to specify a mailbox. I'll go take a look at the URL you posted. |
| 02:24.16 | flenders | when I lower the rx/tx gains to get rid of echo, people can't hear me, or I can't hear people properly |
| 02:24.20 | sudhir492 | I agree echo can be pretty bad with analog cards. However if you have PRI in US, the telco is using echo cancellers anyway. What could be compelling reason to have ec in your hardware then? |
| 02:24.21 | Sweeper | yea, that's what you need |
| 02:24.40 | JT | sudhir492: depends what lines you call |
| 02:24.42 | flenders | so I guess the best option is spend a bit more and get hardware EC |
| 02:24.45 | Sweeper | sudhir492: maybe you're not connecting to the telco |
| 02:24.51 | Zand3r | [TK]D-Fender: Sorry - I have tried "mailbox=200@default" and just "mailbox=200" - both allow message status to be displayed on the phones but neither resolve the issue about not being able to dial out (or the 430 handsets freezing) |
| 02:24.51 | JT | sudhir492: plenty of people in the US swear by hardware EC |
| 02:25.33 | tengulre | anybody working at home in here? |
| 02:25.42 | Sweeper | tengulre: sadly, yes :P |
| 02:25.46 | sudhir492 | Sweeper: what do you mean not connecting to telco? 99% of our calls have a leg on PSTN |
| 02:26.04 | Zand3r | Sweeper: Sorry, jsut to clarify, were you refering to the msg.mwi.1.callBack="200" line I already have or is there something else I need? |
| 02:26.16 | Sweeper | sudhir492: maybe your PRI goes to something other than the pstn |
| 02:26.18 | tengulre | Sweeper: COOL! I like that. |
| 02:26.41 | Sweeper | Zand3r: the subscribe thing needs to be there too, I think |
| 02:27.07 | Sweeper | I use that exact line, it worked the first time *shrug* |
| 02:27.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:27.43 | sudhir492 | Sweeper: No, it is collocated with a telco, and I see a bunch of 5ESS switches, and all kinds of hardcore PSTN stuff :-) |
| 02:27.49 | Zand3r | Sweeper: So you don;t specify a value in the subscribe paramater, you just use it exactly like that URL as I have? |
| 02:28.03 | Sweeper | Zand3r: yes |
| 02:28.14 | tengulre | anybody using jabber in here? I want to buliding a IM, but I don't know which port I need opened in firewall? anybody can give me some tips. |
| 02:28.15 | Sweeper | sudhir492: I meant the hypothetical, generalist "you" :P |
| 02:29.28 | Zand3r | Sweeper: hmm.. thanks for taking the time to clarify. On the one hand that confirms i've at least approached the task the right way but on the other hand completely stumps me as to the problem. I wonder if it could be firmware versions but they are newly purchased phones. |
| 02:29.56 | Sweeper | Zand3r: eh, get the newest firmare from your vendor and stick it on your server anwyas |
| 02:30.04 | Sweeper | it's a good practice anyways |
| 02:30.21 | Sweeper | anways, I think I'm using anyways too often D: |
| 02:30.55 | sudhir492 | I know for sure that the PRIs I have, go through ec at my telcos side. (Once they turned on echo cancellation on D channel when they upgraded their equipment which brought my PRI down and it took them half a day to figure out the reason) |
| 02:30.59 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 02:32.12 | Sweeper | sudhir492: I'm just saying, not everyone's circuits always run through the pstn |
| 02:32.32 | sudhir492 | I agree with you. |
| 02:32.33 | Sweeper | somtimes you *gasp* connect to a channel bank |
| 02:33.00 | Zand3r | Sweeper: Best I can tell I have the latest firmware (1.6.7.0130 on the 501 phones) so it doesn;t appear to be that. |
| 02:33.11 | Sweeper | Zand3r: uh, latest is 2.something |
| 02:33.17 | Sweeper | you hav to get it from your vendor |
| 02:33.39 | sudhir492 | Sweeper: I got it. I had completely forgotten about Channel Bank. Sorry, I misunderstood you. |
| 02:33.53 | Sweeper | no problem :) |
| 02:34.55 | Zand3r | Sweeper: Oh! - My mistake, I had a friend that's here check the polycom website - shows you should always do it yourself. Do you have enough knowledge about the firmware version I have to know if that might be the ptoblem or is it jsut a case of trying and seeing what happens? I can speak to the vendor tomorrow (i'm in the UK) |
| 02:36.26 | Sweeper | Zand3r: the latter, I'm afraid :) |
| 02:36.41 | sudhir492 | Before I went to sleep, request one more time: can anyone provide me the necessary files to program Cisco 7940? I got a few in garage sale :-) and have to program them to be able to use them with Asterisk |
| 02:36.43 | Sweeper | I didn't even try configuration of my phones until I had the new firmware |
| 02:37.05 | Zand3r | Sweeper: Do you mind if I ask what you have in your asterisk sip.conf along with the mailbox= statement |
| 02:37.36 | [TK]D-Fender | SIP 2.1.0 is the latest |
| 02:38.02 | [TK]D-Fender | Zand3r: Your problems are no doubt in your provisioning files |
| 02:38.47 | Sweeper | Zand3r: http://pastebin.ca/391365 |
| 02:39.54 | Zand3r | [TK]D-Fender: Any suggestions and/or examples? All works well right up until I try to have the phones show that there is a mesage. The 501 phones get that right which implies I'm not too far away but then they can;t dial out which I find odd as it seems such an unrelated issue. |
| 02:40.28 | [TK]D-Fender | Zand3r: it IS unrelated |
| 02:40.34 | JT | sudhir492: cisco phones in a garage sale? :o |
| 02:41.06 | [TK]D-Fender | Hey, I've got SpanDSP crashing * on me for a system and need to try and pin down the origin, nothing in *'s logs that I can see. |
| 02:41.35 | Zand3r | [TK]D-Fender: I'll speak to the vendor I purchased the phones from tomorrow to see what the procedure is regarding getting the latest firmware (I checked before purchasing that they were polycom approved and could therefore supply new firmware as it is released) - It's just frustrating havign to wait so I may as weel eliminate anything else I possibly can. I have not come accross any documented issues with this and my firm |
| 02:41.35 | Zand3r | ware version - no tthat that proves anything. |
| 02:43.12 | sudhir492 | JT: Thats right :-) A small travel agency had service from Covad. The owner is moving out of country and drove to his house to get these phones |
| 02:43.13 | Zand3r | [TK]D-Fender: It can not be totally unrelated as without the changes I make to my asterisk sip.conf file, the phones dial out just fine and when I add "type=friend" and "mailbox=200@default" or equivalent then the message status gets displayed on the phone and I can check voicemail fromthe phone but i can not make calls out through the analogue lines. |
| 02:43.35 | JT | sudhir492: hmm i see |
| 02:43.51 | *** join/#asterisk silvertip257 (n=chatzill@c-71-230-114-9.hsd1.pa.comcast.net) |
| 02:43.56 | [TK]D-Fender | Zand3r: pastebin your sip.conf entries & dialplan |
| 02:44.01 | silvertip257 | goo evening fellas |
| 02:44.05 | silvertip257 | *good |
| 02:44.11 | *** join/#asterisk Cisco_System (n=armenicu@adsl-75-16-40-253.dsl.irvnca.sbcglobal.net) |
| 02:44.34 | Cisco_System | Hi all |
| 02:44.46 | Cisco_System | can anyone help me with some information |
| 02:45.24 | Cisco_System | I need info about what telephon lines do I need |
| 02:45.27 | silvertip257 | ask your question(s) and if we know we will tell ay |
| 02:45.30 | silvertip257 | *ya |
| 02:46.15 | silvertip257 | Does anyone know of a compiled article or list of businesses/organizations that are using some form of Asterisk? |
| 02:46.19 | Cisco_System | Like I need 200 phone lines |
| 02:46.47 | Cisco_System | or more , and I want Internet channel to use with them, |
| 02:47.04 | Cisco_System | can you help me what to chose? |
| 02:47.11 | JT | you want some PRIs |
| 02:47.32 | silvertip257 | Cisco_System: have you calculated Erlangs yet? |
| 02:47.45 | Sweeper | mmm, erlang |
| 02:47.54 | Sweeper | damn but I want the company to get me certed in that |
| 02:48.05 | Sweeper | instant 10k raise, much? |
| 02:48.07 | Sweeper | :3 |
| 02:48.12 | Cisco_System | I just need to know what I need, Like I will need some cisco switch for 200 lines |
| 02:48.54 | Cisco_System | silvertip257 what is Erlang ? |
| 02:48.58 | JT | Sweeper: as in the language? |
| 02:49.00 | Cisco_System | I'm new in this :) |
| 02:49.03 | silvertip257 | oh |
| 02:49.08 | [TK]D-Fender | Cisco_System: Sangoma A108d - 8 port PRI card |
| 02:49.31 | JT | or even better, 2 or them and an L1 failover box |
| 02:49.42 | silvertip257 | Cisco_System you calculate Erlangs to figure out how much bandwidth you need for a VoIP setup |
| 02:49.47 | Sweeper | JT: well, and ericsson switches, of course |
| 02:49.56 | silvertip257 | then you can find out what lines you need |
| 02:49.58 | Sweeper | however, I'm gonna suspect Cisco_System is trolling |
| 02:50.01 | JT | s/voip/phone/ :) |
| 02:50.15 | Sweeper | since he's pretending ESL and is using SBC |
| 02:50.22 | Sweeper | and joined #asterisk with the nick Cisco_System |
| 02:50.34 | Cisco_System | come on |
| 02:51.04 | Cisco_System | I just need some info , for the better way is that I can save more money |
| 02:51.08 | Cisco_System | :) |
| 02:51.16 | Sweeper | nice try :P |
| 02:51.28 | Cisco_System | silvertip257 can you help me with calculation ? |
| 02:51.36 | silvertip257 | sure I could |
| 02:51.41 | xpot | can anyone help me with the features.conf? Do I need to add something to extensions.conf in order to use features? I currently have the following: exten => _NXXNXXXXXX,n,Set(DYNAMIC_FEATURES=hangup#play#testfeature#automon) |
| 02:51.41 | xpot | exten => _NXXNXXXXXX,n,Dial(IAX2/blah@provider/1${EXTEN},,L(${RMS}[:60000])W) |
| 02:51.42 | Zand3r | [TK]D-Fender: http://pastebin.ca/391379 |
| 02:51.54 | Sweeper | besides, that last bit of fake ESL was off |
| 02:51.54 | silvertip257 | let's go to a private chat Cisco_System |
| 02:52.02 | JacksLivr | man i am getting so close. when i take the handset off-hook the 7910 reboots. any ideas why? |
| 02:52.09 | Zand3r | [TK]D-Fender: Should numberplan-custom-1 be mentioned in my sip.conf ? |
| 02:52.10 | Cisco_System | tnanks |
| 02:52.52 | [TK]D-Fender | Zand3r: .... I don't do GUI support.. wrong channel |
| 02:53.30 | JT | esl? |
| 02:53.35 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 02:53.38 | Sweeper | english second language |
| 02:53.49 | Zand3r | [TK]D-Fender: I am not using the GUI - I installed using AsteriskNow so I didn;t have to compile Asterisk and dependencies, etc. and I added my users origianlly through the gui but everything has been done directly in the config files since then. |
| 02:54.01 | xpot | when I hit #9 during the call monkeys do not play |
| 02:54.28 | xpot | and *1 does not record... I think I am missing something that allows features to work. |
| 02:55.48 | [TK]D-Fender | Zand3r: Your phones are using [default]. I don't SEE [default] in there, now do I? |
| 02:56.03 | *** join/#asterisk ShoyuRamen (n=ShoyuRam@CPE-67-48-104-5.natnow.res.rr.com) |
| 02:58.31 | Zand3r | [TK]D-Fender: Sorry - I have updated at http://pastebin.ca/391388 |
| 02:59.29 | [TK]D-Fender | Zand3r: Congratulations, you can dial 200. And thats ALL |
| 02:59.52 | Zand3r | [TK]D-Fender: It might be my lack of understanding but incoming calls route through [DID_trunk_1] which I pass to [incoming] and this works fine. Outgoing calls go out through [numberplan-custom-1] and this also works fine but not when I make the changes to sip.conf |
| 03:00.16 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:00.50 | [TK]D-Fender | Zand3r: Outgoing doesn't pass through anything. your phone uses a context, and can dial what IT gives the phone. Your [default] context allows them to dial 200, and nothing more |
| 03:02.07 | *** join/#asterisk silvertip257 (n=chatzill@c-71-230-114-9.hsd1.pa.comcast.net) |
| 03:02.52 | Zand3r | [TK]D-Fender: I am obviously very confused about contexts - I appologise if you feel I am wasting your time - the phones can however dial each other so more than 200 but perhaps that is standard and performed outside of the scopeof the contexts? |
| 03:03.28 | [TK]D-Fender | Zand3r: 201 can only dial 200. |
| 03:03.54 | [TK]D-Fender | Zand3r: And by that I mean the obvious call to check for VM |
| 03:05.39 | *** part/#asterisk ShoyuRamen (n=ShoyuRam@CPE-67-48-104-5.natnow.res.rr.com) |
| 03:05.50 | Zand3r | [TK]D-Fender: When the context is "default" in the sip.conf I promsie I can dial 202 from 201 and have the phone allocated to 202 ring. |
| 03:06.21 | Zand3r | [TK]D-Fender: But, if I change the ccontext in sip.conf to "numberplan-custom-1" thenI can dial out as well which is obviously what you have been implying |
| 03:07.13 | [TK]D-Fender | Zand3r: Not quite, but you are getting the idea. You really need to read up on how contexts work to segragate dialing access. Go download the nice free book. |
| 03:07.15 | [TK]D-Fender | ~book |
| 03:07.25 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:07.48 | r0d3nt | rename it to verizon.. if things continue down this path... |
| 03:08.08 | r0d3nt | " Verizon ! the patent holder for all things packet/voice related ! " |
| 03:08.24 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 03:08.28 | Zand3r | So, [TK]D-Fender You obviously wouldn;t believe it but I am using the book - I'm jsut that bad |
| 03:09.12 | [TK]D-Fender | Zand3r: Nevere said you were bad, just assumed you were unequiped\ |
| 03:09.31 | [TK]D-Fender | Zand3r: Keep at it. You need to learn the very foundation stuff, and its not that hard |
| 03:10.02 | [TK]D-Fender | Zand3r: I'm a little more brief with you because I'm working on 2 computers right now, and IRC is on my laptop.... |
| 03:11.16 | Zand3r | [TK]D-Fender: I really appreciate the time you ahve taken to point me in the right direction and I will of course stick at it. I've not been doing this too long (although I have gone grey in the process) btu I have been really impressed and pleased with the results so far. |
| 03:11.55 | [TK]D-Fender | Zand3r: Glad to hear. This SHOULD take a little time to sort out and as you grow, we should be here to help |
| 03:14.08 | ltdwk | If that's you being brief I'd had to see the verbose you |
| 03:14.13 | ltdwk | s/had/hate |
| 03:20.40 | Zand3r | [TK]D-Fender: Well, you appear to have solved my problem ! - I can now do eveything I'd expect to be able to do on the Polycom 501 phones and they are flashing away to inform me they have messages. The 430 phones still freeze but that could very well be a firmware issue. I still need to read up on contexts in more detail than I evidently have already but I relaly want to say thank you for takign the time to get me over thi |
| 03:20.40 | Zand3r | s hurdle. |
| 03:22.48 | xpot | can anyone help with features? |
| 03:26.33 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 03:26.33 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 03:34.32 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 03:34.41 | *** join/#asterisk cervi2 (n=paceit00@rhlx02.hs-esslingen.de) |
| 03:34.55 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 03:35.59 | cervi2 | can asterisk 1.4 be used in a productive environment or should I use 1.2? |
| 03:37.45 | ltdwk | i'd stick with 1.2 for the time being |
| 03:38.08 | cervi2 | ol |
| 03:38.09 | cervi2 | ok |
| 03:38.15 | ltdwk | or.... test your rig with 1.4 and make sure it works before you put it in production |
| 03:38.16 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:38.50 | cervi2 | the reason I ask is, that a friend of me uses 1.4 and has nearly every day a CPU load of 100% |
| 03:40.01 | cervi2 | maybe I should try with 1.2 |
| 03:41.57 | ltdwk | 1.2 is very stable |
| 03:42.41 | ltdwk | System uptime: 19 weeks, 2 days, 3 hours, 51 minutes, 56 seconds |
| 03:43.23 | *** part/#asterisk abooker (n=tb@c-24-19-226-67.hsd1.mn.comcast.net) |
| 03:52.54 | *** join/#asterisk bmg505 (n=leon@c1-236-5.rndf.isadsl.co.za) |
| 03:55.15 | *** part/#asterisk cervi2 (n=paceit00@rhlx02.hs-esslingen.de) |
| 03:57.26 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:01.13 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 04:01.25 | Mahmoud | do we have any asterisk developers here? |
| 04:02.03 | file | why do you ask? |
| 04:02.19 | *** part/#asterisk silvertip257 (n=chatzill@c-71-230-114-9.hsd1.pa.comcast.net) |
| 04:02.53 | Mahmoud | regarding channel protocols, specially chan_sip.c, it's better if they don't print static names by mentioning evey time. it's better to use #define instead |
| 04:03.16 | Mahmoud | for example. I had to change the sip header from SIP/2.0 into SXP/2.0. I had to change every line that included this.. |
| 04:03.28 | Mahmoud | if they were using #define, it was just a matter of changing a single line |
| 04:03.36 | Mahmoud | this is just an exampe. there are more.. |
| 04:03.51 | file | why are you changing stuff like that? |
| 04:04.09 | gambolputty | he mentioned his ISP censoring thing |
| 04:05.02 | Mahmoud | yeah, my ISP blocks SIP based on patterm match |
| 04:05.10 | gambolputty | and what country are you in? |
| 04:05.16 | Mahmoud | but any how, it's a good programming habit to use #define for such thing |
| 04:05.22 | file | there are just some things that should never be changed under normal circumstances... |
| 04:05.25 | Mahmoud | gambolputty, UAE, single isp |
| 04:06.18 | Mahmoud | instead of printing "SIP/2.0" every time, it's better to just print SIPVR as defined by #define |
| 04:06.38 | Mahmoud | make the code easier to modify |
| 04:06.42 | gambolputty | with an altered SIP header going out, can you use * then? |
| 04:06.51 | Mahmoud | sadly, i'm a network guy rather than a programmer.. i wish to help you and do it my self |
| 04:07.09 | Mahmoud | gambolputty, yep, I patched eyeBeam, so it also sends/recieves modified headers |
| 04:07.34 | *** join/#asterisk Strom_C (n=pocketir@m115e36d0.tmodns.net) |
| 04:07.44 | Strom_C | hi |
| 04:07.48 | Mahmoud | the way my ISP gladly patterm matches SIP is because it's very easy to match it by deep inspection.. |
| 04:07.48 | gambolputty | does the UAE filter IAX traffic too? |
| 04:07.50 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 04:07.53 | Mahmoud | gambolputty, no |
| 04:08.03 | gambolputty | give it time, give it time :( |
| 04:08.09 | Mahmoud | lol |
| 04:08.18 | gambolputty | what else they censor? |
| 04:08.25 | Mahmoud | all proxies |
| 04:08.36 | Mahmoud | anything at 8080 is redirected to their proxy |
| 04:08.44 | Mahmoud | no proxy (port 80 tcp) is also redirected to them |
| 04:08.54 | gambolputty | web traffic |
| 04:08.59 | Mahmoud | yeah |
| 04:09.03 | gambolputty | rich |
| 04:09.07 | gambolputty | what about email? |
| 04:09.10 | Mahmoud | we are not allowed to have web servers in da house |
| 04:09.26 | Mahmoud | we cand send emails |
| 04:09.32 | Mahmoud | but didn't try to run my own mail server |
| 04:09.42 | gambolputty | you would probably have to use PGP at that point |
| 04:09.54 | Mahmoud | man.. |
| 04:10.08 | JacksLivr | http://pastebin.ca/391429 || When I dial the skinny ext, the phone reboots and i get kicked outta the console session. |
| 04:10.09 | Mahmoud | pgp has nothing to do with the port number |
| 04:10.21 | gambolputty | I mean email encryption |
| 04:10.21 | Mahmoud | heh |
| 04:10.53 | Mahmoud | gambolputty.. who cares to read emails sent from a miserable guy's house server |
| 04:11.09 | gambolputty | if they are going to forbid SIP, is scanning emails that much of a stretch for them? |
| 04:11.11 | gambolputty | probably not |
| 04:11.25 | xpot | anyone know how to use features? |
| 04:11.35 | Mahmoud | that's too much.. gambolputty.. how old are you |
| 04:11.39 | gambolputty | 41 |
| 04:11.48 | gambolputty | why? |
| 04:12.16 | Mahmoud | gambolputty, stop watching too much of science fiction movies |
| 04:12.40 | JT | it isn't science fiction |
| 04:12.46 | gambolputty | all I am saying is that one type of censorship would probably lead to another |
| 04:12.46 | JT | it's science fact |
| 04:12.49 | JT | reading emails is easy |
| 04:13.05 | Mahmoud | why on earth why would ISPs read my emails to track if i'm using SIP or not |
| 04:13.13 | gambolputty | and how old are you Mahmoud? |
| 04:13.18 | Mahmoud | it's easy, but time is expensive |
| 04:13.27 | Mahmoud | 23 years old |
| 04:13.45 | gambolputty | you might as well look into a VPN and/or SIP/IAX encryption |
| 04:13.51 | JT | they might just scan them, and i don't think he specifically meant to look for sip talk |
| 04:13.58 | Mahmoud | a young guy from the middle of the desert dared to insult a 41 yo europian man :P |
| 04:14.11 | JT | gambolputty: srtp doesn't hide the fact it's sip |
| 04:14.31 | gambolputty | use a VPN then |
| 04:14.38 | gambolputty | like OpenVPN |
| 04:14.45 | Mahmoud | why sould I use vpn and all the bloated features |
| 04:14.49 | Mahmoud | i use openvpn for other stuff |
| 04:14.57 | Mahmoud | for SIP, modify the header and all work happy dandy |
| 04:15.05 | JacksLivr | Would someone please take a look at this? Thanks. http://pastebin.ca/391429 || When I dial the skinny ext, the phone reboots and i get kicked outta the console session. |
| 04:15.08 | Mahmoud | no need to waste extra b/w for features you don't need |
| 04:15.22 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:18.31 | Mahmoud | well, i hope that developers hear this "use more of #define rather than printing static strings in chan_sip.c every time" |
| 04:20.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:20.36 | Corydon76-home | Mahmoud: for what? |
| 04:21.00 | JunK-Y | static srings are changing sometimes. |
| 04:21.26 | Mahmoud | i'm talking about things that don't change |
| 04:22.12 | Mahmoud | instead of saying (for example) printf("%s","SIP/2.0"); let's say printf("%s",SIPVR); where SIPVR is defined by #define |
| 04:23.08 | JT | don't you mean printf("SIP/2.0"); for the first one? |
| 04:23.40 | Mahmoud | heh yeah.. i'm not programmer |
| 04:23.48 | Mahmoud | sorry for the funny example, but i hope it's clear |
| 04:23.57 | JT | i understand what you mean |
| 04:24.13 | *** join/#asterisk PhilMcKracken (n=chatzill@adsl-11-183-176.mia.bellsouth.net) |
| 04:24.22 | Mahmoud | i used a bit of C years ago.. but it was for monthes only, as i'm only a networking guy |
| 04:25.54 | Corydon76-home | I'm not sure that's a good idea, as we may want to specify different levels of compliance to the SIP spec, based on whatever SIP command we're using at the time |
| 04:26.30 | JT | well surely the original chan_sip would remain compliant if no-one changed the #define |
| 04:26.58 | JT | i mean, there's nothing stopping people randomly change stuff set in strings throughout the code, it's just annoying |
| 04:27.15 | Corydon76-home | JT: what I mean is that a single define may not express the gamut of what we want to express in the code |
| 04:27.28 | Corydon76-home | It's best to keep the version bound to the particular command |
| 04:27.40 | Kumbang | hi guys, does anyone experience connecting Atlas 550 with * through T1 PRI |
| 04:27.56 | JT | what other things would you want to use where SIP/2.0 is normally used? |
| 04:28.07 | Corydon76-home | Possibly SIP/1.0 |
| 04:28.24 | Corydon76-home | or SIP/2.1 or SIP/3.0 at some point |
| 04:28.39 | JT | well that could be changed with a define or some logic code and two #defines |
| 04:28.47 | JT | i'm not seeing how static strings are easier to change |
| 04:29.24 | Corydon76-home | I'm not seeing how making it a define is necessary |
| 04:29.49 | Corydon76-home | and I can see reasons why it would be a bad idea |
| 04:29.55 | JT | necessary, debatable, advantageous, probably |
| 04:30.00 | file | it's not, there's no valid reason you need to change it under normal circumstances |
| 04:30.03 | JT | especially for weird scenarios like Mahmoud's |
| 04:30.31 | Mahmoud | it's just a cleaner way of doing things. it allows changing thigs easier, less errors..etc |
| 04:30.45 | JT | just because something is not normally changed, doesn't mean there isn't advantages to using a define in some cases |
| 04:31.03 | Corydon76-home | You're welcome to submit a patch on the bugtracker |
| 04:31.10 | JT | it's not my issue :) |
| 04:31.28 | Mahmoud | i wish i could, but i'm not a programmer.. |
| 04:31.29 | JT | i don't have the nazi isp ;) |
| 04:31.31 | Corydon76-home | but I'll warn you ahead of time, you need to make a case for why you think the patch should go in |
| 04:31.47 | Mahmoud | a patch for cleaning things up |
| 04:31.52 | JT | Mahmoud: just find and replace all instances? |
| 04:32.05 | Mahmoud | JT, I just did, and sip no longer blocked |
| 04:32.06 | Corydon76-home | Like I said, you'll need to make a case for it |
| 04:32.18 | JT | Mahmoud: oh, it definately works? |
| 04:32.23 | Mahmoud | JT, yeah |
| 04:32.32 | Qwell | Mahmoud: for how long is sip no longer blocked? |
| 04:32.38 | JT | Mahmoud: umm then it's easy to make a patch, use diff |
| 04:32.41 | Mahmoud | JT, one more thing, DTMF is not sent, something else other than SIP.. SIP is 100% working |
| 04:32.45 | file | Qwell! |
| 04:32.48 | Qwell | file! |
| 04:32.53 | Qwell | we got lost |
| 04:32.55 | file | Qwell: I missed you <3 |
| 04:32.58 | Qwell | eep! |
| 04:33.03 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqut.cable.mindspring.com) |
| 04:33.07 | JT | Mahmoud: what dtmf mode? |
| 04:33.12 | Qwell | What did I miss? |
| 04:33.25 | wunderkin | you missed file? |
| 04:33.25 | file | Qwell: the end of the world |
| 04:33.29 | Qwell | ahh |
| 04:33.50 | Mahmoud | JT, the default one, (inband works, but not with GSM that i use), info doesn't work.. i guess they are doing further pattern match to detect tones |
| 04:33.59 | Qwell | today was actually pretty eventful |
| 04:34.01 | JT | Mahmoud: no |
| 04:34.03 | Mahmoud | i can stablish calls, but can not send DTMF during calls |
| 04:34.08 | JT | Mahmoud: inband only works with g.711 |
| 04:34.22 | Mahmoud | JT, i know, i said i use GSM, so it doesn't work with me |
| 04:34.23 | Corydon76-home | eventful or uneventful? |
| 04:34.28 | JT | you must use INFO or rfc2833(?) if not using g.711 |
| 04:34.28 | Qwell | eventful |
| 04:34.35 | Mahmoud | JT, yeah |
| 04:34.36 | JT | Mahmoud: that's got nothing to do with your evil isp |
| 04:34.44 | Mahmoud | JT, so? |
| 04:34.45 | JT | gsm cannot accurately carry dtmf indications |
| 04:34.52 | JT | inband |
| 04:34.55 | Mahmoud | JT, i know.. i'm not talking about inband |
| 04:34.56 | file | just use RFC2833 |
| 04:35.09 | Mahmoud | holy cow.. did i say GSM is not working with inband because of my ISP? |
| 04:35.30 | file | make sure dtmfmode is set too, or else your DTMF might be wonky |
| 04:35.31 | JT | < Mahmoud> JT, the default one, (inband works, but not with GSM that i |
| 04:35.31 | JT | <PROTECTED> |
| 04:35.31 | JT | <PROTECTED> |
| 04:35.39 | Mahmoud | good, read what i said |
| 04:35.45 | JT | you're the one who said pattern matching |
| 04:35.47 | JT | chill out |
| 04:35.54 | Mahmoud | JT, i was talking about "info" mode |
| 04:36.05 | JT | try rfc then |
| 04:36.15 | Mahmoud | rfc doesn't work, tried |
| 04:36.17 | JT | and make sure the other side supports whatever mode you choose |
| 04:36.31 | Mahmoud | eyebeam supports info and rfc, none of them work |
| 04:36.43 | Mahmoud | it's 100% my ISP |
| 04:36.48 | JT | you may need to check things out at a lower level |
| 04:36.52 | Mahmoud | i have to modify more of the chan_sip.c to let info working |
| 04:36.53 | file | RFC2833 travels over the same stream as audio |
| 04:37.11 | Mahmoud | file, i know, but can be patterm matched since it carries DTMF. right? |
| 04:37.25 | file | it's binary |
| 04:37.30 | Qwell | well, if they did that, they'd just filter the voice too |
| 04:37.31 | Mahmoud | pattern matching that binary |
| 04:37.32 | JT | until you see that the packets are being transmitted from asterisk, and then see they are not being received at the other end, using a packet sniffer, you cannot say 100% it is your isp |
| 04:37.49 | Mahmoud | Qwell, voice data, and DTMF content is different |
| 04:37.54 | file | yes... you need to debug the issue further |
| 04:37.58 | Qwell | not really |
| 04:38.02 | Qwell | it's all just rtp |
| 04:38.13 | Mahmoud | Qwell, same headers, but different content |
| 04:38.31 | Mahmoud | Qwell, the content is the DTMF number we are sending.. |
| 04:38.50 | file | just use RFC2833 and do an rtp debug, if you see it... then your ISP is not the issue |
| 04:39.27 | Mahmoud | i'll do it but 100% won't see it |
| 04:39.34 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 04:39.44 | JT | now who's the one being paranoid about the isp? |
| 04:39.47 | Mahmoud | because, i RFC works in my own network, but once passes the internet (ISP) it doesn't |
| 04:40.06 | JT | there could be another issue at play |
| 04:40.21 | Mahmoud | such as? |
| 04:40.36 | JT | i just don't understand why they wouldn't destroy all rtp comms instead of just dtmf? |
| 04:40.51 | Mahmoud | RTP can be used by other stuff than voip? |
| 04:41.14 | Mahmoud | actually, they didn't even block port UDP 5060, since it could be used by other apps (dynamic ip range) |
| 04:41.22 | Mahmoud | s/ip/port/ |
| 04:41.35 | JT | if they kill sip, doesn't sound like they like rtp anyway |
| 04:41.47 | Mahmoud | sure, but RTP is used by other stuff as well |
| 04:42.01 | Mahmoud | they are serving large amount of customers, the whole country! |
| 04:43.04 | Mahmoud | anyway, they are morons.. |
| 04:43.23 | Mahmoud | block voip :/ they are evil |
| 04:45.26 | Mahmoud | just a typical issue when the internet providers is the phone provider.. |
| 04:45.37 | Mahmoud | s/providers/provider/ |
| 04:46.15 | JT | and there's no other providers... |
| 04:46.19 | Mahmoud | nope |
| 04:46.53 | Mahmoud | there is another one, but very new and limited to a single city only, with a very bad service, and only about mobile phones (not even wired phones, or the internet) |
| 04:47.21 | Mahmoud | they have very bad services regarding their signals |
| 04:51.08 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 04:53.25 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 04:53.34 | Strom_C | catsex |
| 04:53.56 | Qwell | dogballs |
| 04:54.09 | Corydon76-home | dead hookers |
| 04:54.16 | Strom_C | muahahaha |
| 04:54.41 | russellb | goatse |
| 04:55.02 | Strom_C | russellb is not trained in the art of catsex |
| 04:55.07 | sevard | nope |
| 04:55.23 | sevard | you're going to a catsex free hell |
| 04:55.34 | russellb | that's .... a good thing, i thin |
| 04:55.48 | sevard | wrongobucko |
| 04:55.52 | russellb | i lost my cellphone within a few minutes of getting home |
| 04:55.54 | russellb | wtf |
| 04:56.02 | sevard | what's your # |
| 04:56.07 | russellb | 17 |
| 04:56.14 | sevard | what's the rest |
| 04:56.21 | Strom_C | 6 |
| 04:56.25 | Strom_C | is 6 taken? |
| 04:56.25 | sevard | nice. |
| 04:56.25 | russellb | 6 ... niner ... |
| 04:56.27 | russellb | alpha ... |
| 04:57.31 | sevard | beta max |
| 04:57.45 | JacksLivr | wow, looks like the gang has assembled |
| 04:58.05 | russellb | ok, seriously, someone call my phone please :( |
| 04:58.07 | Strom_C | alpha bravo catsex dogballs echo foxtrot golf |
| 04:58.14 | sevard | russellb: paste your damn digits |
| 04:58.15 | Strom_C | russell: ill dial |
| 04:58.23 | russellb | sevard: i don't know you :-p |
| 04:58.26 | sevard | i know you. |
| 04:58.30 | Qwell | stalker |
| 04:58.45 | sevard | it's called 'distant loving' |
| 04:58.57 | JacksLivr | you go to jail for that |
| 04:59.01 | Corydon76-home | hotel india juliet kilo lima mike november oscar papa quebec romeo sierra tango uniform victor x-ray yankee zulu |
| 04:59.02 | russellb | i found it! |
| 04:59.07 | russellb | it made it under the bed somehow ... |
| 04:59.10 | russellb | thanks Strom_C |
| 04:59.31 | Corydon76-home | Oops, missed whiskey |
| 04:59.44 | Strom_C | russell: find it? |
| 04:59.55 | russellb | yup |
| 04:59.56 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:00.22 | russellb | nice voicemail :) |
| 05:00.28 | Strom_C | hahaha thanks |
| 05:00.32 | sevard | was it "doggggggggggggballllllllllllllzzzz" |
| 05:00.41 | JacksLivr | what else was under there? |
| 05:00.44 | russellb | pretty much |
| 05:00.48 | russellb | JacksLivr: nothing :) |
| 05:01.08 | JacksLivr | 7910 problem almost fixed |
| 05:01.15 | Qwell | JacksLivr: what's the problem? |
| 05:01.21 | JacksLivr | i am sooooo close |
| 05:01.39 | JacksLivr | now when i call that extension or pick up the 7910 handset, it reboots |
| 05:01.53 | JacksLivr | and sometimes kicks me outta the console |
| 05:01.58 | sevard | Guys |
| 05:02.06 | Strom_C | tesla girls |
| 05:02.07 | JacksLivr | Strom has made me paranoid about spelling |
| 05:02.13 | sevard | I forgot where we are dining tonight |
| 05:02.50 | sevard | anyone? |
| 05:03.00 | sevard | lame. |
| 05:03.13 | Strom_C | taco bell telephone company |
| 05:03.43 | sevard | Strom_C binrev>asterisk |
| 05:04.02 | Strom_C | shrug |
| 05:04.14 | Strom_C | i can only be on one network from my mobile phone |
| 05:04.48 | sevard | well, you missed the-awesome |
| 05:05.01 | Strom_C | what happened? |
| 05:05.11 | sevard | not telling |
| 05:05.13 | sevard | nahnahnahnah |
| 05:05.28 | Strom_C | well then ill read it when i get back to the hotel |
| 05:05.34 | Strom_C | then ill kick your ass |
| 05:05.41 | sevard | bring it on beochhhh |
| 05:09.56 | *** part/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 05:12.03 | *** join/#asterisk zillao (n=medinaor@201.240.24.130) |
| 05:12.54 | JacksLivr | wow, the word beochhhh really brought on the silence |
| 05:22.52 | boch | how can i recognize the channel type of ${CHANNEL} ? |
| 05:24.28 | Strom_M | use CUT and GotoIf() statements |
| 05:27.40 | boch | thanks |
| 05:39.07 | *** join/#asterisk p4r14h (n=j0sh@69.92.145.178) |
| 05:40.23 | boch | Strom_C, btw, just found ${CHANNEL(channeltype)} |
| 05:40.36 | Strom_M | oh! |
| 05:40.43 | Strom_M | i didnt realize it was a function |
| 05:41.08 | Qwell | Strom_C: I expect you to wear the "nub zone" sign tomorrow |
| 05:41.22 | Qwell | s/<tab>/<tab><tab>/ |
| 05:41.26 | Strom_M | hehehe |
| 05:45.08 | Strom_M | that still exists? |
| 05:45.19 | Strom_M | and do I have to wear te entire traffic cone too? |
| 05:46.29 | Qwell | of course - that's part of the sign |
| 05:46.36 | Qwell | last I saw it...was...hmm |
| 05:46.46 | Qwell | I don't remember where, but it was there about 2 weeks ago |
| 05:47.03 | Qwell | I should buy a mini nub-zone hat |
| 05:47.13 | Qwell | one of those small plastic ones, heh |
| 05:48.39 | JacksLivr | you should all go to bed. Strom you need to look happy in the morning. |
| 05:48.51 | Qwell | bed? It's only 1 |
| 05:49.01 | JacksLivr | are you a vampire? |
| 05:49.07 | Qwell | I'm a developer |
| 05:49.23 | Qwell | take that as you will |
| 05:49.36 | JacksLivr | id be in bed but i hate to let somehting beat me. this phone *WILL* ring tonight! |
| 05:50.00 | Qwell | using 1.4? |
| 05:50.09 | Qwell | and I assume chan_skinny? |
| 05:50.44 | JunK-Y | Qwell: any reason why dial.c has been comitted to 1.4.1, since this is new feature? |
| 05:50.52 | Qwell | JunK-Y: to fix a bug |
| 05:51.05 | Qwell | it's only used in app_page and maybe SLA I believe |
| 05:51.32 | JunK-Y | ha the new SLA stuff, good point. |
| 05:51.48 | Qwell | I'm not even sure if that uses it |
| 05:51.49 | JacksLivr | <GULP> using 1.2 |
| 05:51.49 | Qwell | I know app_page does though - that's why it was written |
| 05:52.06 | Qwell | JacksLivr: upgrade to 1.4 if you're using chan_skinny... you'll have *far* better results |
| 05:52.15 | JunK-Y | k, didnt take a look at that new stuff, will do later this week. |
| 05:52.21 | Qwell | I rewrote a considerable portion of it |
| 05:52.55 | JacksLivr | ok, when i installed someone told me to go 1.2 |
| 05:53.12 | JacksLivr | im going to go through my logs and kill somebody |
| 05:54.07 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 05:54.19 | sumasuma | i do get a segfault when i run asterisk |
| 05:54.39 | sumasuma | no changed from the downloaded source given in asterisk.org/download |
| 05:54.40 | Qwell | sumasuma: Please post a detailed report at bugs.digium.com |
| 05:54.45 | JacksLivr | 21:39 < JacksLivr> why 1.2 over 1.4 |
| 05:54.45 | JacksLivr | 21:51 < JT> JacksLivr: 1.2 is more stable, less bleeding edge |
| 05:55.10 | Qwell | well, for chan_skinny, you want bleeding edge ;) |
| 05:55.17 | JacksLivr | i guess so |
| 05:55.32 | JacksLivr | ok, this phone will not ring tonight |
| 05:55.47 | Qwell | I should dig out my 7910... |
| 05:55.54 | Qwell | I'm out of room on my desk though |
| 05:56.16 | JacksLivr | do you have a favorite wifi phone you use? |
| 05:56.31 | Qwell | 7920, heh |
| 05:56.36 | Qwell | wifi phones suck though, in general |
| 05:56.55 | Qwell | actually, from what I hear, the Kirk sets (sold in EU, and I don't think they even sell them anymore) are really cool |
| 05:57.02 | Qwell | base station with DECT handsets |
| 05:57.11 | Qwell | runs on sccp |
| 05:57.25 | JacksLivr | i have a 7920 that i use at work. how hard would it be to jump back and forth to call manager at work and * at home |
| 05:57.47 | JT | JacksLivr: why are you quoting me from what i assume is a while back? |
| 05:57.58 | Qwell | not very - the phones read the config at boot |
| 05:58.05 | ltdwk | he spoke the truth too |
| 05:58.30 | JacksLivr | JT: you did not give bad advice |
| 05:58.38 | JT | good to hear ;) |
| 05:58.45 | JacksLivr | you appropriately disclaimed |
| 05:59.10 | JacksLivr | i didnt know i needed bleeding edge back then |
| 05:59.20 | JT | neither did i |
| 05:59.44 | JacksLivr | i'll learn to upgrade tomorrow |
| 06:00.04 | JacksLivr | dont think i wanna tackle that right now |
| 06:01.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-8-247.red.bezeqint.net) |
| 06:02.34 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:03.55 | JacksLivr | ima bed tiem now plees! thanks for everything. Qwell, thanks for the 7910/1.4 advice. nite all! |
| 06:04.44 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:04.49 | PhilMcKracken | Qwell: Why do Wifi Phones suck? |
| 06:05.01 | Qwell | PhilMcKracken: they just do... there aren't really any good ones |
| 06:05.18 | PhilMcKracken | Bad reception? Interface? |
| 06:05.24 | Qwell | all of the above |
| 06:05.45 | PhilMcKracken | Anything's better than a cell phone |
| 06:05.47 | Qwell | battery life |
| 06:05.47 | JT | PhilMcKracken: jitter, variable lag, speed scaling |
| 06:05.52 | JT | PhilMcKracken: unlikely |
| 06:05.54 | Qwell | AP switching |
| 06:06.12 | PhilMcKracken | qwell: Do you use a softphone then? |
| 06:06.16 | Qwell | nah |
| 06:06.22 | Qwell | I use a cellphone :P |
| 06:06.36 | PhilMcKracken | with voip? |
| 06:06.39 | Qwell | no |
| 06:06.46 | Qwell | I don't voip for the sake of voip |
| 06:07.12 | Qwell | I mean, think about it.. when are you ever going to be in a situation where there isn't a phone more than 20 feet from you? |
| 06:07.18 | PhilMcKracken | Then What do you use asterisk 4 |
| 06:07.24 | Qwell | and, where a wifi phone would work |
| 06:07.47 | Qwell | heh, I don't |
| 06:08.27 | Qwell | I just like making phones ring :P |
| 06:09.07 | PhilMcKracken | You're an op on asterisk who doesn't even use it? |
| 06:09.19 | Qwell | don't have to use it to be a developer |
| 06:09.28 | Qwell | I do it because I enjoy writing code, and like I said - making phones ring |
| 06:09.35 | Qwell | I *hate* actually talking on phones |
| 06:10.15 | PhilMcKracken | You're obviously insane. So JT, but do you think sound quality is inherently worse w/ VoIP |
| 06:10.33 | Qwell | PhilMcKracken: obviously ;) |
| 06:11.16 | Qwell | as far as sound quality goes, it's more the hardware itself.. ulaw is equiv to what's used on traditional phone lines |
| 06:11.35 | Qwell | but, with wifi, you also introduce jitter, lag, things like that |
| 06:11.53 | ltdwk | the same things you introduce on a cell phone |
| 06:11.53 | Qwell | a cellphone has the same problems, but it masks those things really well |
| 06:12.19 | ltdwk | i prefer to use ATA's with analog handsets for wireless applications |
| 06:12.38 | Qwell | ltdwk: yep |
| 06:12.45 | ltdwk | range is better and you generally don't get jitter just static |
| 06:13.00 | PhilMcKracken | ATA's? |
| 06:13.03 | Qwell | ~ata |
| 06:13.05 | jbot | it has been said that ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 06:13.25 | PhilMcKracken | i see |
| 06:13.34 | Qwell | you get the best of both worlds there |
| 06:13.51 | PhilMcKracken | Can you use an ATA with an old skool rotary phone? |
| 06:13.54 | Qwell | and if you get a good cordless handset, the range is about the same as wifi |
| 06:13.59 | Qwell | yep, and I do |
| 06:14.02 | PhilMcKracken | Now that would sooooo be worth it |
| 06:14.21 | Qwell | I have an old Stromburg_C rotary phone with an iaxy inside |
| 06:14.27 | Qwell | Strom_C: :P |
| 06:14.45 | ltdwk | pimping |
| 06:14.49 | JT | the range is much further with a cordless handset, than wifi |
| 06:14.54 | ltdwk | i bet the chicks drool over that thing. |
| 06:14.54 | Qwell | bright ass orange too :D |
| 06:15.12 | Qwell | ltdwk: my wife saw it at a consignment store, and wanted it, so we bought it, and I stole it :P |
| 06:15.17 | Qwell | $3! |
| 06:15.26 | Qwell | I *could not* pass that up |
| 06:15.50 | Qwell | it's a beast of a phone too - weighs a good 5-6 lbs |
| 06:15.56 | yonahw-work | can someone please look at this pastebin and tell me why my voicemail is not playing the instructions? http://pastebin.ca/391533 |
| 06:16.03 | PhilMcKracken | qwell: so what's yer number? |
| 06:16.15 | Qwell | PhilMcKracken: I don't have one - just my cellphone and work |
| 06:17.34 | Mahmoud | heh |
| 06:17.42 | Qwell | 8... |
| 06:17.47 | Mahmoud | operators are free to poke any one, but can we poke them? :/ |
| 06:17.52 | Qwell | and my house isn't wired for telephone, heh |
| 06:17.53 | JT | i bet you don't have Motorola desktop consoles :P |
| 06:17.57 | JT | they look like phones |
| 06:18.04 | Mahmoud | Qwell, all voip? |
| 06:18.04 | JT | but they're not |
| 06:18.06 | Qwell | JT: I have Motorola Phone Tools - does that count? |
| 06:18.15 | ltdwk | Qwell: if no telephone, then no DSL? |
| 06:18.17 | JT | nah, these are ex police radio |
| 06:18.18 | Qwell | Mahmoud: nah, 4 voip, plus 1 that's technically voip |
| 06:18.21 | Qwell | ltdwk: cable |
| 06:18.36 | JT | they use a special line |
| 06:18.38 | Qwell | I voiped my rotary phone, for the sake of voip :P |
| 06:18.41 | JT | 4 wire thing |
| 06:18.45 | JT | hah |
| 06:18.51 | Mahmoud | Qwell, didn't get you. 4 voip (aren't these technically voip?) |
| 06:18.54 | JT | that's a stromism |
| 06:18.56 | Qwell | (note: i do not advise that) |
| 06:19.05 | Qwell | JT: oh, he was jealous of my phone |
| 06:19.11 | JT | heh |
| 06:19.11 | Qwell | he may not admit it - but he was |
| 06:19.19 | JT | i've got a 5 line analogue phone |
| 06:19.24 | Qwell | Mark saw it, and was thrilled :D |
| 06:19.25 | JT | it's a funny thing |
| 06:19.33 | Qwell | it's Digium orange :D |
| 06:19.39 | PhilMcKracken | qwell: Do you use your cell with VoIP? |
| 06:19.44 | Qwell | PhilMcKracken: nope |
| 06:19.49 | ltdwk | i've got one line coming into an x100p at my home with snom handsets |
| 06:20.06 | Qwell | well |
| 06:20.10 | Qwell | that's not true |
| 06:20.14 | Qwell | technically.. |
| 06:20.26 | Qwell | I have my cell connected to asterisk via chan_bluetooth :D |
| 06:20.30 | JT | i've got limited telephony at home, only a T1 channel bank and softphones |
| 06:20.36 | ltdwk | the x100p was from that OpenVox mob |
| 06:20.38 | Qwell | JT: how minimalist |
| 06:20.44 | JT | yeah |
| 06:20.51 | Mahmoud | living in cities with 802.11 wireless networks all over the area is cool.. we can use voip wireless phones any where |
| 06:21.10 | Qwell | Mahmoud: okay, that's probably the only benefit of a wifi phone I can think of |
| 06:21.13 | ltdwk | JT: A PRI at home ? |
| 06:21.15 | denon | Mahmoud: sure, as long as you don't mind everyone else listening to your calls |
| 06:21.26 | JT | ltdwk: no, a T1 card and T1 channel bank, not a pri |
| 06:21.40 | Mahmoud | denon, heh? they can be encrypted.. and btw, your current analog cell phone is _no_ encrypted either |
| 06:21.42 | PhilMcKracken | denon: And you can't listen in to cell phone calls? |
| 06:21.47 | ltdwk | JT: define 'T1 channel bank' |
| 06:21.51 | JT | analog cell phone? rofl |
| 06:22.03 | Mahmoud | JT, yeah |
| 06:22.04 | denon | Mahmoud: you're running wep on open networks? |
| 06:22.17 | JT | PhilMcKracken: they can't, most cell calls are digital and encrypted |
| 06:22.18 | Mahmoud | denon, wep sucks, wpa |
| 06:22.21 | denon | cell phones can easily be encrypted |
| 06:22.34 | denon | Mahmoud: are you talking a city-wide service? or some schmuck with an open linksys? |
| 06:22.36 | JT | do many people still use analogue in the US? |
| 06:22.40 | Mahmoud | denon, i'm not running anything.. just saying it would be nice to have 802.11 wirelss in large cities |
| 06:22.46 | Mahmoud | denon, city-wide |
| 06:22.47 | denon | JT: most areas dont even have analog signal anymore |
| 06:22.48 | JT | there is no unencrypted cell calls in .au |
| 06:22.54 | JT | that's what i thought |
| 06:22.57 | PhilMcKracken | denon: Who's encryption? |
| 06:23.07 | Mahmoud | denon, and even if it was wep-based, then it's fine. we can secure SIP communications |
| 06:23.17 | JT | the telcos', it protects from over the air interception |
| 06:23.19 | denon | PhilMcKracken: I believe it's in cdma spec |
| 06:23.39 | denon | and yes, I know it all terminates to clear channel somewhere |
| 06:23.51 | PhilMcKracken | Backdoored! |
| 06:23.53 | JT | yeah usually the PRIs in the ground |
| 06:23.54 | denon | but I trust my cell company's oc48 more than I trust some guy with an open linksys |
| 06:24.26 | Qwell | denon: Did you just use the words "trust" and "cell company" in the same sentence? :) |
| 06:24.28 | Mahmoud | denon, any case, can't SIP send encrypted voice data? |
| 06:24.37 | denon | Qwell: certinaly, mine is locally owned :) |
| 06:24.38 | JT | ltdwk: a channel bank converts a T1 in CAS RBS mode to lots of analogue lines |
| 06:24.38 | PhilMcKracken | Yes it can |
| 06:24.49 | JT | Mahmoud: SRTP can send encrypted rtp payloads |
| 06:24.50 | denon | Mahmoud: sure, if your phone and pbx supports it |
| 06:24.51 | ltdwk | JT: Oh I see |
| 06:24.55 | JT | sip signalling remains unencrypted |
| 06:25.01 | Qwell | denon: ir(just to piss off Strom_C)regardless. ;) |
| 06:25.08 | Mahmoud | denon, so, no worries about joining some open linksys networks ;) |
| 06:25.15 | ltdwk | JT: So you interface using t1 signalling but the calls go out analog lines? |
| 06:25.17 | PhilMcKracken | Mahmoud: and *you* get to decide how |
| 06:25.28 | Mahmoud | PhilMcKracken, huh? |
| 06:25.30 | denon | Mahmoud: well, it's also illegal |
| 06:25.32 | JT | ltdwk: that's right, it's not a pri because there's no D channel |
| 06:25.38 | denon | and you're telling them exactly who you are, by your PBX's ip |
| 06:25.38 | Mahmoud | denon, i think in the US only |
| 06:25.44 | denon | anywhere, I'd imagine |
| 06:25.49 | PhilMcKracken | Mahmoud: and *you* get to decide what you use to excrypt the SIP call |
| 06:25.56 | denon | it's unauthorized access, and theft of service |
| 06:26.12 | denon | just because you leave your house door open, doesn't mean you're asking people to come steal your tv |
| 06:26.13 | ltdwk | JT: I loosely use the term pri to describe any kind of channel bank (E1, t1) |
| 06:26.18 | Mahmoud | denon, oh, you were talking about stealing wirelss from neighbores |
| 06:26.29 | denon | from whoever |
| 06:26.36 | JT | ltdwk: PRI describes only an interface using isdn common channel signalling |
| 06:26.57 | JT | to be technically accurate |
| 06:26.58 | Mahmoud | denon, i thought you meant just "open" wireless networks that do not require any encryption/authentication |
| 06:27.18 | denon | nah, I mean some schmuck who buys a linksys, opens the box, jacks it in, and thinks everything is good |
| 06:27.21 | ltdwk | JT: yeah i know, but... |
| 06:27.26 | Mahmoud | heh |
| 06:27.44 | Mahmoud | some people live by these schmucks |
| 06:28.38 | PhilMcKracken | denon: I think there's a big difference between wireless service and physical property in someone's home |
| 06:28.45 | Qwell | bed time |
| 06:29.09 | denon | PhilMcKracken: so if I find an exploit in your web server, and use it to deliver my uber cool spam, that's ok? |
| 06:29.21 | Qwell | http://asterisk.org/node/48327 |
| 06:29.22 | PhilMcKracken | if someone takes your tv, you can't use it at all |
| 06:29.25 | Qwell | for anyone who missed it earlier ;D |
| 06:29.50 | PhilMcKracken | there's a difference between that and wireless service too |
| 06:30.30 | PhilMcKracken | I mean, I've run open access points. It's not uncommon to do |
| 06:32.59 | PhilMcKracken | I've never thought it okay for random people to come into my home |
| 06:33.21 | yonahw-work | can anyone help me with the voicemail() command? I can not get my voicemail to play the instructions, when it picks up it immediately plays the beep. |
| 06:33.22 | yonahw-work | http://pastebin.ca/391533 |
| 06:38.30 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:38.41 | *** join/#asterisk Strom_C (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 06:39.38 | ltdwk | yonahw-work: are you presenting either the u or b flag, and if not, are you sure that the sound files are avaialable ? |
| 06:41.16 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 06:42.04 | *** join/#asterisk czaveri (n=czaveri@203.193.154.194) |
| 06:48.08 | yonahw-work | i am not presenting the u or b flag |
| 06:48.38 | yonahw-work | i did not record any outbound greetings if i use the flags do they present standard greetings? |
| 06:49.53 | yonahw-work | i will try that |
| 06:51.11 | yonahw-work | itdwk: thanks, that fixed the problem |
| 07:00.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:02.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 07:04.16 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 07:10.20 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:17.11 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:19.28 | *** join/#asterisk appelza (n=pieter@196.40.106.74) |
| 07:19.50 | *** part/#asterisk appelza (n=pieter@196.40.106.74) |
| 07:27.42 | Vec | Does anyone know about the Grandstream BT-102, giving trouble with DHCP, I 12 of them and they all seem erraticaly get an ip address ? |
| 07:32.17 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:33.40 | gfraysse | <PROTECTED> |
| 07:34.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:34.11 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.14.14.revip2.asianet.co.th) |
| 07:34.42 | BugKhaM | is it possible to add sip user with no password? |
| 07:34.51 | BugKhaM | haven't tried it |
| 07:36.41 | Braxus | older firmware versions for the BT-102 supposedly have DHCP issues. |
| 07:37.01 | *** join/#asterisk lorinc (n=ang@pool-2896.adsl.interware.hu) |
| 07:37.29 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 07:41.48 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:46.24 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:56.44 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:05.49 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:06.35 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:23.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:36.27 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 08:40.45 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:47.10 | *** join/#asterisk abuyazan (n=khaled@dogbert.palnet.com) |
| 08:47.38 | abuyazan | hi aLL |
| 08:48.29 | abuyazan | uwe, hi |
| 08:48.37 | abuyazan | uwe, can you help me in asterisk |
| 08:48.42 | abuyazan | uwe, hahahahaa |
| 08:49.19 | uwe | :) |
| 08:51.35 | *** part/#asterisk null3d (n=xokappa@69.46.31.46) |
| 08:51.58 | uwe | in reality :) |
| 08:59.00 | uwe | thank you jeremy_g :) |
| 09:02.04 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 09:04.27 | jeremy_g | :) |
| 09:05.14 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:06.23 | *** join/#asterisk Merlin83b (n=Merlin83@office.34sp.com) |
| 09:06.39 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:07.58 | uwe | um, well ... in the full log, the number beside DEBUG and VERBOSE denotes a uniqe number for a call, right? |
| 09:08.09 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 09:08.58 | uwe | heh ... i just locked abuyazan out ... hes seeking revenge |
| 09:09.09 | *** join/#asterisk Strom_M (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 09:09.09 | *** part/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 09:09.19 | uwe | abuyazan, ;) |
| 09:10.36 | russellb | uwe: no, i think that is the thread ID ... not terribly useful, really |
| 09:11.11 | Strom_M | damn it, I can't sleep |
| 09:11.52 | uwe | im getting music on hold inturrupting calls sometimes, and im unable to know what triggers this behaviour !! |
| 09:13.06 | russellb | Strom_M: same here |
| 09:14.52 | Strom_M | gah |
| 09:14.59 | Strom_M | I have to teach tomorrow! |
| 09:16.15 | *** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 09:16.51 | russellb | Strom_M: :-/ |
| 09:19.20 | key2 | russellb |
| 09:22.51 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 09:22.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:23.59 | russellb | i'm running out of ideas to feed my insomnia |
| 09:24.17 | Strom_M | wikipedia |
| 09:24.21 | Strom_M | that'll last you days |
| 09:27.34 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 09:32.05 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 09:37.20 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:38.29 | tparcina | 5038, is that port of asterisk managment interface or something else? |
| 09:40.40 | RoyK | by default, that's the manager, yes |
| 09:41.39 | Inez | yeah |
| 09:42.58 | tparcina | RoyK: can I use telnet, ssh, or something else to connect and type commands to AMI? |
| 09:42.58 | Nugget | telnet is eeeeeeevil! |
| 09:43.33 | tparcina | Nugget: I know that, I'm just checking what are options I have :) |
| 09:45.21 | russellb | yes, you can use telnet to the manager interface ... |
| 09:45.32 | russellb | though it's not terribly useful for more than testing |
| 09:45.49 | russellb | there are easier ways to do things from the CLI ... |
| 09:45.57 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 09:46.15 | RoyK | tparcina: use telnet |
| 09:46.31 | RoyK | Nugget: ever tried to use ssh to debug the manager? |
| 09:47.02 | tparcina | RoyK: russellb: so, when I use telent, someone will be able to sniff packets and see my manager username/pass, right? |
| 09:47.15 | angryuser | can router-modem option 'block udp flood or block tcp flood' disturb the Qualigy of sip peers? |
| 09:47.23 | angryuser | *Quality |
| 09:47.49 | angryuser | mdr i mean qualify option set to yes;) |
| 09:48.01 | RoyK | tparcina: of course. the manager doesn't have any encryption for that sort of stuff |
| 09:48.08 | RoyK | tparcina: so use an ssh tunnel or something |
| 09:48.12 | RoyK | stunnel works too |
| 09:48.20 | RoyK | if you're sending the user/pass over a wan |
| 09:49.07 | russellb | it supports SSL in trunk ... |
| 09:49.13 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 09:49.19 | russellb | but that's probably not what you are using :) |
| 09:49.32 | RoyK | russellb: I somehow doubt someone will use trunk in production :) |
| 09:49.44 | russellb | for some crazy reason ... |
| 09:49.52 | RoyK | :P |
| 09:50.44 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 09:50.51 | tparcina | RoyK: thank you |
| 09:52.21 | RoyK | np :) |
| 09:55.50 | angryuser | has anyone been using wondershaper to prioritize SIP straffic? |
| 09:55.51 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:55.53 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:56.06 | angryuser | *traffic |
| 09:56.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 09:58.26 | lesouvage | I used to pick a timestamp with SetVar(TIME=${DATE}${TIMESTAMP}). Tis doens't seem to work in 1.4. what is the new way to get a timestamp? |
| 09:59.18 | russellb | first, i don't think SetVar is still in 1.4. |
| 09:59.22 | russellb | it is Set |
| 10:00.35 | lesouvage | The variables are still the same? |
| 10:03.56 | lesouvage | This should do? Set(TIJD_STEMPEL=${DATE}_${TIME}) |
| 10:07.27 | *** join/#asterisk gmaxwell (n=gmaxwell@wikimedia/KatWalsh/x-0001) |
| 10:08.08 | gmaxwell | http://bugs.digium.com/print_bug_page.php?bug_id=9027 < I'm hitting this problem with asterisk 1.4 but 1.2 was fine. Should I reopen this bug? |
| 10:14.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:14.39 | *** join/#asterisk Mavvie (n=edwin@ppp16-84.lns2.syd7.internode.on.net) |
| 10:16.48 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 10:16.48 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:29.51 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 10:36.45 | *** join/#asterisk stony (n=steinche@p57B3A032.dip0.t-ipconnect.de) |
| 10:36.51 | stony | hi |
| 10:37.01 | zoa | ho |
| 10:37.19 | stony | i can't find a list of webinterfaces for asterisk on asterisk.org - am I blind or is there no list ? |
| 10:37.29 | zoa | there is no list there, try voip-info.org |
| 10:37.37 | stony | thx |
| 10:38.47 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 10:39.22 | *** join/#asterisk Corydon76-home (i=peuce@pdpc/supporter/sustaining/Corydon76-home) |
| 10:39.22 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 10:41.12 | *** join/#asterisk psychobilly (n=Fuzz@zaf.on-line-cafe.gr) |
| 10:41.39 | psychobilly | is there support for zrtp or srtp in asterisk? |
| 10:44.25 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 10:45.45 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:48.41 | zoa | no |
| 10:51.23 | psychobilly | any idea of a free pbx that supports any of these? |
| 10:58.34 | stony | are there any free web interfaces for asterisk available ? |
| 10:58.48 | psychobilly | freebpx |
| 10:59.25 | psychobilly | freepbx i mean :) |
| 10:59.57 | stony | that's a whole distro isn't it ? |
| 11:00.08 | psychobilly | no |
| 11:00.34 | psychobilly | u are talkink about trixbox trixbox |
| 11:00.42 | stony | ah ic |
| 11:00.45 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:01.43 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 11:01.52 | tzafrir | interesting: asterisk -rx reload # does not reload logger settings here |
| 11:02.08 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 11:02.12 | tzafrir | on a fairly recent 1.2 |
| 11:02.18 | tzafrir | Is this a feature? |
| 11:02.41 | stony | on freepbx.org you can download the screenshots as "plain text" - what a feature ;) *G* |
| 11:03.11 | tzafrir | stony, aalib versions? |
| 11:03.30 | stony | hmm no |
| 11:03.45 | stony | just a misplaced format-link |
| 11:08.33 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 11:11.46 | tparcina | how do I execute SIPpeers command in AMI? I'm connected to AMI with telnet. |
| 11:12.01 | tparcina | I have sucesfuly loged in |
| 11:15.18 | tparcina | I found it - I only need to send this command: "Action: SIPpeers" |
| 11:15.34 | tparcina | this is nice :)) I like AMI!!! ;) |
| 11:24.13 | *** join/#asterisk Corydon76-home (i=peuce@pdpc/supporter/sustaining/Corydon76-home) |
| 11:24.13 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:25.38 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
| 11:30.10 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 11:39.31 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 11:42.17 | *** join/#asterisk Strom_M (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 11:43.56 | *** join/#asterisk omri (n=omri@omri.linuxil.org) |
| 11:44.17 | omri | hey |
| 11:44.31 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381998.dsl.bell.ca) |
| 11:47.14 | omri | I'm having trouble activating the http server for asterisk. I've edited httpd.conf and defined enabled=yes; I've also changed manager.conf and added a user, as well as defining enabled=yes and webenabled=yes; am I missing something? |
| 11:47.45 | omri | also note that I'm accessing the webserver from a computer on the LAN via http://10.0.0.1:8088 |
| 11:47.56 | omri | I'm getting a 404 not found error |
| 11:48.02 | LeddyHM | you guys have to do anything special for DST, my phones aren't showing the corrent time and they are ntp'd to the server, which shows the correct time |
| 11:48.09 | LeddyHM | restart asterisk, ntpd? |
| 11:48.47 | florz | LeddyHM: NTP doesn't know anything about timezones. |
| 11:48.49 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 11:49.50 | LeddyHM | hrm, I wonder where the disconnect is then |
| 11:51.55 | *** join/#asterisk shazaum_ (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 11:52.01 | omri | nvm, I had to access a praticular form |
| 11:52.16 | shazaum_ | somebody??? |
| 11:53.15 | tparcina | AMI, DBPut doesn't work like it's stated in http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+DBPut |
| 11:53.25 | tparcina | on Asterisk 1.2.15 |
| 11:53.44 | shazaum_ | somebody "cisco 7960 ip phone" ? |
| 11:53.46 | tparcina | anybody knows what is the right command? |
| 11:54.12 | tparcina | shazaum_: I have it, it works fine on SIP |
| 11:54.48 | shazaum_ | tparcina: I do not obtain to bring up to date firmware for sip |
| 11:54.52 | shazaum_ | do you help me ? |
| 11:54.55 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 11:55.09 | shazaum_ | i use tftp and cisco tftp |
| 11:55.20 | tparcina | shazaum_: sorry, I don't understand you |
| 11:55.35 | shazaum_ | mmm |
| 11:55.40 | shazaum_ | tparcina: moment |
| 11:55.41 | shazaum_ | :D |
| 11:55.46 | shazaum_ | my english is bad |
| 11:55.47 | tparcina | ok :) |
| 11:56.04 | tparcina | shazaum_: It will be good enough :) |
| 11:56.15 | tparcina | I'm not perfect nihter |
| 11:58.27 | shazaum_ | :( |
| 11:58.27 | shazaum_ | ok |
| 11:58.35 | shazaum_ | i have a Cisco 7960 |
| 11:58.42 | shazaum_ | IP phone |
| 11:59.05 | tparcina | ok, i get that |
| 11:59.13 | shazaum_ | i have that to make upgrade |
| 11:59.30 | shazaum_ | but I do not obtain |
| 12:00.36 | shazaum_ | make one upgrade I use tftp server |
| 12:01.07 | tparcina | anybody, why does AMI command DBPut doesn't work for me - http://pastebin.ca/391789 |
| 12:01.33 | tparcina | shazaum_: do you have cisco firmware on your computer? |
| 12:01.41 | shazaum_ | yes |
| 12:02.09 | tparcina | and you have tftp server up and running with that cisco image in root of that tftp? |
| 12:02.27 | shazaum_ | yes |
| 12:02.52 | shazaum_ | file P0S3-08-6-00 |
| 12:03.00 | tparcina | and you have setup the phone to get image from that tftp server? |
| 12:03.14 | shazaum_ | yes |
| 12:03.27 | shazaum_ | in menu "TFTP1" |
| 12:03.47 | tparcina | and when you tourn on the phone, does it tries to download that image from tftp server? |
| 12:04.33 | shazaum_ | yes |
| 12:04.41 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 12:05.05 | shazaum_ | I see this in logs |
| 12:05.41 | shazaum_ | but.... |
| 12:06.23 | shazaum_ | when phone makes download |
| 12:06.23 | shazaum_ | t only catches 2 archives |
| 12:06.38 | shazaum_ | it only catches 2 archives |
| 12:07.48 | tparcina | shazaum_: use this link, it has helped me to put my first firmware on cisco phone - http://www.loligo.com/asterisk/Cisco/79xx/upgrading.79xx.phones |
| 12:07.54 | shazaum_ | tparcina: you understand me ? |
| 12:08.50 | tparcina | yes, you need to edit some files so that your phone knows what firmware he should download. I'm don't rememeber what files you should edit, but look at the link that I have sent you |
| 12:09.50 | shazaum_ | yes...yes... |
| 12:10.13 | shazaum_ | I edited the archives |
| 12:10.22 | shazaum_ | OS79XX.TXT |
| 12:10.41 | shazaum_ | P003-08-6-00 > OS79XX.TXT |
| 12:14.21 | _charly_ | hi |
| 12:14.34 | LeddyHM | found it. It was in sip.cfg the start/end of dst |
| 12:15.36 | *** join/#asterisk tuxick (n=userMurf@a82-95-232-97.adsl.xs4all.nl) |
| 12:15.37 | tuxick | lo |
| 12:16.03 | DrukenLPY | morning peoples |
| 12:16.11 | shazaum_ | morning |
| 12:16.16 | *** join/#asterisk Psykick (n=anon@60-234-168-153.bitstream.orcon.net.nz) |
| 12:16.18 | Psykick | hi guys |
| 12:16.56 | shazaum_ | tparcina: you have firmware of the 7960 IP phone? |
| 12:16.57 | Psykick | I've having a few issues with getting my 2 x TDM400P with 2 FXS and 2 FXO modules in one card and 4 FXO modules in the other working |
| 12:17.01 | shazaum_ | Psykick: hi :) |
| 12:17.11 | Psykick | hi shazaum_ |
| 12:17.43 | Psykick | I'm using TB 2.0 (not by choice) and have run genzaptelconf that comes with TB 2.0 |
| 12:18.31 | DrukenLPY | what is TB ? |
| 12:18.41 | Psykick | in card 1 the modules are 1: FXS 2: FXS 3: FXO 4: FXO |
| 12:18.51 | Psykick | in card 2 the modules are 1-4: FXO |
| 12:19.35 | Psykick | now whenever I place a call to an external number Zap/1-1 constantly rings and the destination never rings at all |
| 12:19.39 | Psykick | TB = TrixBox |
| 12:20.52 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:21.02 | DrukenLPY | Psykick: i doubt you'll find any help here... you should try trixbox's support channel |
| 12:21.03 | Psykick | I've checked the signalling which is set to fxs_ks for all the fxo modules and fxo_ks for all the fxs modules |
| 12:21.33 | Psykick | DrukenLPY: should it really matter .... I'm just trying to get the zap channels working |
| 12:22.02 | Psykick | but in the off chance .... |
| 12:22.10 | Psykick | are they on the same irc network? |
| 12:22.31 | DrukenLPY | yeah in #freepbx |
| 12:22.43 | tuxick | why does "Read the docs online!" give me pdf.zip only? |
| 12:22.54 | Psykick | cool thanks DrukenLPY |
| 12:25.04 | *** join/#asterisk coppice (n=chatzill@74.197.17.210.dyn.pacific.net.hk) |
| 12:27.37 | Psykick | great ... no one in #freepbx is awake |
| 12:27.50 | Merlin83b | Not a patient man then, Psykick? |
| 12:28.04 | Psykick | I am actually very patient |
| 12:28.18 | Merlin83b | 4 minutes and then quitting the channel doesn't sound that patient. |
| 12:28.24 | Psykick | I just typed a message asking if anyone was actually reading what I had typed and received no answer |
| 12:28.39 | JT | most people ignore those messages on purpose |
| 12:29.09 | Psykick | I don't wanna get into any arguements ... but purposefully not answering those isn't very helpful |
| 12:29.40 | JT | a question asking if anyone is there isn't a question asking for real help... |
| 12:29.59 | Psykick | no ... but it is just asking for a simple response |
| 12:30.22 | Psykick | a hi .. or hello is sufficient |
| 12:30.48 | JT | i'm just trying to explain it, you can continue with your expectations :) |
| 12:30.48 | Psykick | I will apologise in advance if I do offend anyone ... |
| 12:31.02 | Psykick | I'm just at my teathers end at the moment |
| 12:31.08 | Psykick | and am seriously frustrated |
| 12:31.36 | Psykick | as one can imagine |
| 12:31.42 | Psykick | JT: lol |
| 12:32.06 | JT | maybe your ports are in the wrong order? |
| 12:32.30 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 12:32.32 | Psykick | JT: that's what appears to be the issue |
| 12:32.57 | Psykick | its like the FXS channels are answering the calls for the FXO channels |
| 12:33.36 | JT | so change it? |
| 12:33.47 | FreezeS | problem is that it's in the egg |
| 12:34.12 | Psykick | I've gotta disconnect from IRC for a bit in order to place a call |
| 12:34.17 | Psykick | I need to VPN to the box |
| 12:34.47 | Psykick | only select network traffic is allowed ... but I'll try connecting anyway while connected to IRC to see if I lose the connection |
| 12:35.41 | JT | egg? |
| 12:36.20 | FreezeS | the good thing is that it's a folder |
| 12:36.27 | FreezeS | I thought it was a compressed file |
| 12:40.00 | stony | this freepbx thing is even more circular then editing the configs by hand |
| 12:40.52 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:50.47 | *** join/#asterisk Strom_C (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 12:53.05 | angryuser | stony: simple configs pbx >>web more difficult>> standalone |
| 12:53.13 | *** join/#asterisk taishi (n=juanma@zion.dattaweb.com) |
| 12:53.16 | taishi | Hello everyone |
| 12:53.31 | taishi | How can I know what codec is a client is trying to use when reaching my * server? |
| 12:54.00 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:54.27 | angryuser | taishi: sip show peer <peer> |
| 12:55.32 | TaiSHi | angryuser: Will that work even if he can't make calls (it's my ITSP) because of "No compatible codecs' ? |
| 12:56.32 | angryuser | TaiSHi: no |
| 12:57.52 | angryuser | TaiSHi:in 'Codec Order' you wont have any info, try them one by one in sip.con 'allow=<codec>' |
| 12:57.53 | TaiSHi | Great :P |
| 12:58.05 | TaiSHi | Mmm |
| 12:58.26 | TaiSHi | On his ext is clearly specified "allow=all" |
| 12:59.00 | angryuser | TaiSHi:try to specify them, who knows;) |
| 12:59.04 | TaiSHi | ok |
| 12:59.21 | TaiSHi | Got any list ? :P |
| 13:00.01 | angryuser | TaiSHi:alaw,ulaw,g729, >> core show codecs |
| 13:00.17 | Dr-Linux | again forgot |
| 13:00.38 | TaiSHi | Ok, great |
| 13:00.43 | TaiSHi | angryuser: regarding g729 |
| 13:00.50 | Dr-Linux | TaiSHi: how can i do this patch? i mean in which extension i should save this patch? |
| 13:00.50 | Dr-Linux | http://bugs.digium.com/view.php?id=9245 |
| 13:00.57 | TaiSHi | If my provider pays license, do I have to pay it aswell? |
| 13:01.10 | angryuser | TaiSHi:yes |
| 13:01.30 | TaiSHi | Goddamn provider, gonna burn him in hell ¬¬ |
| 13:01.47 | TaiSHi | Dr-Linux: Guess it's a .diff ... |
| 13:01.56 | TaiSHi | Honestly, extensions don't matter in linux |
| 13:02.40 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:02.45 | Dr-Linux | well, when i open the patch i gets open as code, so i copy and paste the code in notepad, but what extension it should have? :S |
| 13:02.47 | TaiSHi | Still, I haven't applied a patch in years :P |
| 13:02.58 | angryuser | TaiSHi: you can try precompiled g729 (open 729) |
| 13:03.09 | angryuser | Dr-Linux: .patch |
| 13:03.18 | Dr-Linux | ok good |
| 13:03.18 | TaiSHi | angryuser: Where could I find such a great thing? |
| 13:03.26 | Dr-Linux | and what should be the file name? |
| 13:03.33 | TaiSHi | Dr-Linux: You picked the wrong person asking me about windows patching u_U |
| 13:03.37 | TaiSHi | .exe is the way. |
| 13:03.39 | Dr-Linux | exactly as in attached file? |
| 13:03.44 | TaiSHi | Yes, probably |
| 13:03.52 | Dr-Linux | lol |
| 13:03.53 | Dr-Linux | heheh |
| 13:04.13 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:04.14 | Dr-Linux | TaiSHi: that was just an example |
| 13:04.25 | Dr-Linux | ofcos it's on linux and on my production server |
| 13:04.32 | Dr-Linux | i gonna patch to skinny chan |
| 13:05.24 | TaiSHi | Ok |
| 13:05.25 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:05.33 | TaiSHi | I'm gona google for open 729 |
| 13:05.54 | TaiSHi | angryuser: May I love you ? |
| 13:05.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:06.13 | TaiSHi | So you're not angry anymore <3 |
| 13:06.48 | DrukenLPY | that's some tough love |
| 13:06.49 | Dr-Linux | lol |
| 13:06.57 | TaiSHi | Yawn |
| 13:06.59 | Dr-Linux | what's exact command to patch? |
| 13:07.04 | Dr-Linux | p0 ? :S |
| 13:07.06 | Dr-Linux | or something |
| 13:09.24 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 13:10.05 | angryuser | TaiSHi: google for open g.729 i dont remember where it is, but it dont give yoy the right to use it in certain countries where G729 licensing apply |
| 13:10.12 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 13:10.33 | Dr-Linux | angryuser: can you tell me the exact command to patch? |
| 13:10.58 | nettie | Hi guys, I have an hfc based isdn adapter, what should I use as driver please? chan_capi or zaphfc? which one is supposed to be more stable? thanx in advance. |
| 13:12.32 | angryuser | Dr-Linux: oh, well first you must determine where ptch has to be placed, if it is relative or not, then 'man patch' |
| 13:13.13 | Dr-Linux | angler: it's in /usr/asterisk/ ... to the chan_skinny.c |
| 13:13.20 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 13:13.21 | angryuser | nettie: i have used visdn and chan_capi had no particular problems |
| 13:14.04 | JT | nettie: zapfhc/bristuff |
| 13:14.48 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 13:15.13 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 13:15.13 | nettie | JT what are the difference between the 2? |
| 13:15.45 | nettie | angryuser well the fact you didnt have problems is still something -- thanx :P |
| 13:16.05 | Dr-Linux | damn |
| 13:16.15 | Dr-Linux | can anybody tell me the exact patch command? |
| 13:16.33 | angryuser | nettie: try the bristaffed version, autoinstall easy one |
| 13:16.40 | Dr-Linux | patch file should be executale? |
| 13:16.41 | angryuser | *brisstuffed |
| 13:16.53 | nettie | http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1p.tar.gz ?? |
| 13:17.19 | nettie | will go for the bristuff-0.3.0-current.tar.gz |
| 13:17.29 | angryuser | nettie: current |
| 13:17.48 | angryuser | Dr-Linux: ses 'man patch' |
| 13:17.53 | angryuser | *see |
| 13:18.12 | Dr-Linux | accordin to that i should do what i pasted above |
| 13:18.14 | Dr-Linux | -PBX asterisk]# patch -p0 asterisk-trunk-7935.patch |
| 13:18.25 | Dr-Linux | but there is nothing |
| 13:21.14 | Dr-Linux | ever anyone patched to any asterisk chan/app ? :S |
| 13:21.48 | angryuser | Dr-Linux: where the patch file is placed? |
| 13:22.08 | Dr-Linux | angryuser: on the currently directory |
| 13:22.33 | angryuser | Dr-Linux: and the patch file is relative to what directory? |
| 13:23.08 | angryuser | Dr-Linux: it has absolute or relative path? |
| 13:23.16 | Dr-Linux | /usr/src/asterisk/channes/chan_skinny.c |
| 13:24.04 | TaiSHi | angryuser: Thanks a lot |
| 13:24.07 | Dr-Linux | angler: i'm not in the .../channels location, but in /usr/src/asterisk |
| 13:24.13 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:24.42 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 13:26.06 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:26.32 | angryuser | Dr-Linux: ask fender he is the geek;) |
| 13:26.39 | TaiSHi | zomg |
| 13:27.11 | Dr-Linux | angryuser: i got it, but looks like my asterisk version is not compitable |
| 13:27.39 | [TK]D-Fender | Stupid DST change.... |
| 13:27.54 | [TK]D-Fender | mangling all my phones, and singular app breakage... |
| 13:28.14 | angryuser | Dr-Linux: at least you know how to apply the patches |
| 13:28.35 | Dr-Linux | i already know that, but this one is not working, |
| 13:28.46 | Dr-Linux | not sure if version issue |
| 13:29.13 | angryuser | Dr-Linux: try the ritual dance, works sometimes |
| 13:29.14 | JerJer | moooo |
| 13:29.36 | Dr-Linux | look here: http://phpfi.com/214814 |
| 13:30.16 | TaiSHi | angryuser: <3 =) (187mb library....) |
| 13:30.26 | angryuser | Dr-Linux:looks like wrong version |
| 13:31.09 | angryuser | TaiSHi:search the web for procompiled ones, you need to place the in /modules that's all |
| 13:31.38 | Dr-Linux | hhm.. |
| 13:31.39 | TaiSHi | Yawn... |
| 13:31.45 | Dr-Linux | angler: tell me something |
| 13:31.46 | TaiSHi | I dont really care downloading... |
| 13:31.52 | TaiSHi | Got 250 MB bandwidth |
| 13:32.00 | Dr-Linux | i'm using asterisk 1.2.1 for zaptel/asterisk/pri |
| 13:32.24 | angryuser | TaiSHi: i look poor with my 8 Mbit;) |
| 13:32.29 | Dr-Linux | but now i want to upgrade only asterisk ? |
| 13:33.03 | TaiSHi | angryuser: This is @ work... @ home I have 640k :'( |
| 13:33.10 | Dr-Linux | angryuser: that would not be a problem if if zaptel is 1.2.1 and asterisk is 1.2.15? |
| 13:33.43 | angryuser | Dr-Linux: nothing is garanteed |
| 13:34.12 | Dr-Linux | :S |
| 13:34.23 | Dr-Linux | i'm asking in normal cases |
| 13:34.41 | angryuser | Dr-Linux: normally it should work |
| 13:34.46 | Dr-Linux | okey |
| 13:34.56 | angryuser | Dr-Linux: in worst case you can allways downgrade |
| 13:35.21 | TaiSHi | Ahh... cant seem to find precompiled |
| 13:35.24 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:35.42 | Dr-Linux | angler: yeah |
| 13:36.10 | Dr-Linux | lemme backup my files |
| 13:38.21 | JT | nettie: the "current" version is usually not current |
| 13:38.21 | JT | look for the highest number/letter |
| 13:38.43 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 13:40.07 | tuxick | any 4xS0 isdn cards that are no problem on linux? |
| 13:40.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:40.49 | angryuser | TaiSHi: found wiki http://kvin.lv/pub/Linux/Asterisk/ |
| 13:40.51 | JT | junghanns cards work fine |
| 13:41.00 | tuxick | checking |
| 13:41.50 | TaiSHi | angryuser: <3 |
| 13:41.50 | tuxick | JT: i guess you mean that quadbri thing? |
| 13:42.39 | JT | tuxick: yes |
| 13:44.47 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.140.56.dyn.user.ono.com) |
| 13:45.05 | zapp-branigan | hi i have this problem Dec 31 16:02:21 NOTICE[391]: iax2-provision.c:515 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. |
| 13:45.28 | zapp-branigan | what is the problem please ? |
| 13:46.03 | TaiSHi | Have you configured IAX provisioning ? |
| 13:46.09 | JT | don't worry about it |
| 13:46.10 | angryuser | zapp-branigan:iaxprov.conf in not present in /etc/asterisk |
| 13:46.11 | tuxick | JT: cheers, noted |
| 13:46.12 | zapp-branigan | hummm no |
| 13:46.14 | JT | unless you have an iaxy |
| 13:46.30 | TaiSHi | Then that may be the answer :) |
| 13:46.42 | JT | it's a NOTICE |
| 13:46.49 | JT | NOTICEs can usually be ignored |
| 13:47.25 | *** join/#asterisk payal (i=payal@ip-209-82-178-130.static.skiplink.net) |
| 13:47.30 | JT | dec 31, why are you showing us an error from 3 months ago? |
| 13:47.30 | payal | hi all |
| 13:47.53 | payal | do Ineed special hardware to ue asterisk |
| 13:47.55 | angryuser | JT: he was isitated |
| 13:48.06 | JT | angryuser: ? |
| 13:48.16 | JT | payal: not if you want to do voip only |
| 13:48.22 | angryuser | JT: isitated to show it;) |
| 13:48.36 | JT | is that even a word? |
| 13:48.54 | angryuser | JT: hm....maybe |
| 13:49.06 | payal | JT: I just want to make a few calls to my bro in Japan on his cell |
| 13:49.22 | JT | payal: you don't necessarily need any hardware |
| 13:49.27 | JT | isitated... |
| 13:49.39 | payal | JT: thanks |
| 13:50.05 | angryuser | <PROTECTED> |
| 13:50.15 | JT | ah |
| 13:50.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:50.37 | JT | i swear he came in with the exact same error a few months ago |
| 13:51.23 | angryuser | JT: sometimes got some language switching delay...do stupid errors |
| 13:51.43 | TaiSHi | Guys, g2g |
| 13:51.46 | TaiSHi | Take care, ttyl |
| 13:51.55 | TaiSHi | Thanks a lot again angryuser |
| 13:53.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:53.37 | tuxick | how much of a learning curve will i face hooking up to isdn when i've only done SIP so far? |
| 13:53.58 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:55.08 | Dr-Linux | Qwell[]: around? |
| 13:57.03 | Dr-Linux | http://bugs.digium.com/view.php?id=9245 |
| 13:57.13 | Dr-Linux | which asterisk version support this patch? |
| 13:57.38 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:01.21 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-f935f548092a54d8) |
| 14:01.24 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 14:01.47 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 14:04.21 | [TK]D-Fender | Dr-Linux: Just a guess but : SVN Revision (number only!) 58557 |
| 14:04.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 14:04.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:05.13 | Dr-Linux | [TK]D-Fender: hhm.. so it's not gonna work with my currenlty deployed asterisk 1.2.1? |
| 14:06.12 | [TK]D-Fender | Dr-Linux: Do I really have to answer this? You you use 2007 Mustang part's on a 1957 classic? |
| 14:06.24 | [TK]D-Fender | Do* |
| 14:06.58 | zoa | is he would have a 1957 classic |
| 14:07.08 | zoa | do you really think he would be here asking about asterisk ? :) |
| 14:07.08 | SuPrSluG | hello all |
| 14:07.15 | Dr-Linux | hhm.. |
| 14:07.44 | Dr-Linux | i think i'd need grab Qwell mind, as he is administrator for this patch and he suggested me to try it |
| 14:07.53 | creativx | i would put a 2007 mustang engine into a 1957 classic |
| 14:07.56 | creativx | or atleast its efi |
| 14:08.21 | SuPrSluG | is it possible to exit voicemail w/out leaving a message and get back to the menu from where you came? |
| 14:08.52 | *** join/#asterisk zeppelin_ (n=zeppelin@201-40-187-171.paemt700.dsl.brasiltelecom.net.br) |
| 14:09.24 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:09.49 | SuPrSluG | i thought pressing # would do it, but it just hangs up |
| 14:10.10 | [TK]D-Fender | SuPrSluG: Only if there's no more dialplan to execute |
| 14:10.35 | [TK]D-Fender | SuPrSluG: You may want to look at the BIG print. |
| 14:10.51 | SuPrSluG | BIG? |
| 14:11.37 | [TK]D-Fender | SuPrSluG: Pastebin your dialplan and the CLI output of your attempt.... |
| 14:12.01 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:14.45 | *** join/#asterisk elisboa (n=elisboa@unaffiliated/elisboa) |
| 14:17.12 | *** join/#asterisk seele_ (n=seele@208.35.116.20) |
| 14:17.33 | seele_ | how can i caonfigure the text2wav (festival) language to spanish??? |
| 14:18.46 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 14:19.28 | sergee | how to disable anonymous SIP calls? |
| 14:19.39 | JerJer | sergee: don't have a default context |
| 14:19.50 | sergee | allowguest=no |
| 14:19.52 | sergee | <PROTECTED> |
| 14:20.44 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 14:21.00 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 14:21.12 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:21.12 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:23.45 | *** join/#asterisk stack_ (n=stack@63.239.190.203) |
| 14:26.51 | seele_ | how can I configure the deafult festival language ???? |
| 14:33.38 | *** join/#asterisk ru_wing (i=wing@82.179.135.4) |
| 14:38.28 | *** join/#asterisk wasabi_ (n=jhaltom@ubuntu/member/wasabi) |
| 14:38.52 | wasabi_ | I don't suppose anybody has any experience using Asterisk in a medium to large call center? I'd like some tutorials or stuff. |
| 14:39.34 | new2345 | does anyone know if AsteriskNow has a script for automatic outbound calling? |
| 14:40.30 | tzafrir | asterisk -rx 'originate WHATEVER' |
| 14:40.44 | *** join/#asterisk TaiSHi (n=juanma@zion.dattaweb.com) |
| 14:40.48 | TaiSHi | Back |
| 14:40.48 | tzafrir | in asterisk as of 1.4 |
| 14:40.54 | TaiSHi | angryuser: downloaded g729 |
| 14:41.02 | TaiSHi | But how do I check if it's correctly loaded? |
| 14:41.02 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:41.32 | angryuser | core show codecs |
| 14:42.01 | new2345 | i meant that would dial a list of numbers, not that i have to write a script for |
| 14:42.11 | TaiSHi | It says it's a list of codecs, but not the actually installed |
| 14:42.26 | angryuser | module show |
| 14:42.28 | TaiSHi | Disclaimer: this command is for informational purposes only. |
| 14:42.28 | TaiSHi | <PROTECTED> |
| 14:43.50 | angryuser | TaiSHi: module show in cli |
| 14:44.20 | TaiSHi | Yep |
| 14:44.22 | TaiSHi | Doing that, thanks |
| 14:44.45 | TaiSHi | format_g729.so Raw G729 data 0 |
| 14:44.45 | *** join/#asterisk CrashSys (n=kumba@bartleby.crashsys.com) |
| 14:44.49 | TaiSHi | ah... pleasure :P |
| 14:44.56 | TaiSHi | Ah... |
| 14:44.57 | TaiSHi | Nooo |
| 14:44.59 | TaiSHi | ¬¬ |
| 14:45.01 | TaiSHi | Damnit |
| 14:45.57 | TaiSHi | Ok, not loading for some reason |
| 14:46.06 | TaiSHi | It loads 'format_g729' |
| 14:46.10 | TaiSHi | But not codec_ ... |
| 14:46.33 | santiago | TaiSHi, show translation |
| 14:46.41 | angryuser | TaiSHi: have you restarted asterisk fter placing module in /modules? |
| 14:46.45 | stack_ | DST is PIMA! :P |
| 14:46.54 | TaiSHi | <PROTECTED> |
| 14:48.45 | angryuser | TaiSHi: try module load 'name' |
| 14:49.23 | TaiSHi | [Mar 12 11:50:02] WARNING[19644]: loader.c:623 load_resource: Module 'codec_g729.so' could not be loaded. |
| 14:49.25 | TaiSHi | yawn |
| 14:50.16 | angryuser | TaiSHi: try to download one optimizet for you CPU |
| 14:50.25 | CrashSys | Polycom sip 2.0 have better DST implementation then 1.6? |
| 14:50.31 | angryuser | *optimised |
| 14:50.37 | CrashSys | Since now it's a second march |
| 14:50.38 | TaiSHi | Got a celeron 1.8... |
| 14:50.48 | TaiSHi | Only AMD/pentium optimized |
| 14:50.58 | TaiSHi | downloading P3 module now |
| 14:51.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:51.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:54.58 | TaiSHi | Wiiiiiiii |
| 14:55.00 | TaiSHi | got it working |
| 14:55.09 | TaiSHi | Just loaded 'other' version |
| 14:56.16 | TaiSHi | angryuser: marry mee |
| 14:57.05 | angryuser | TaiSHi: if you look like Clara Morgane, ok;) |
| 15:02.33 | errr | when I add the sounds tar ball to my asterisk I get make install errors: No description for sounds/phonetic/alpha.gsm make: *** [datafiles] Error 1 any idea why? |
| 15:02.44 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 15:05.37 | errr | basicly I know why it happens, I guess I want to know what to add to the sounds.txt to fix it, or where to get a sounds.txt that has the info about the files that are added by that tar ball |
| 15:11.01 | [TK]D-Fender | CrashSys: Yeah they are poorly layed out. I haven't checked the 2.1.0 addendum to see if they changed it there or not... |
| 15:12.42 | CrashSys | fender: fun fun fun |
| 15:13.36 | [TK]D-Fender | CrashSys: Looks like there is a fix in there... readin up on it now |
| 15:14.02 | [TK]D-Fender | CrashSys: A PROPER fix that is. |
| 15:15.14 | *** join/#asterisk mkl1525 (n=qwertz@pD9532652.dip0.t-ipconnect.de) |
| 15:16.23 | *** join/#asterisk saftsack (n=oliver@p54A7D6DE.dip.t-dialin.net) |
| 15:16.34 | [TK]D-Fender | CrashSys: Confirmed. 2.1.0 is fully compatible for the new rules |
| 15:18.46 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-b4905b711ae3795b) |
| 15:19.36 | CrashSys | So all I have to do is upgrade my polycom's to 2.1.0 and i'm set |
| 15:19.42 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 15:19.44 | CrashSys | and set the rules |
| 15:20.05 | nextime | hello. What's the state of asterisk-gui in svn trunk, it is now working with IE? |
| 15:20.28 | [TK]D-Fender | CrashSys: The provisionig files default to NA's new rules |
| 15:24.15 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:25.44 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:26.38 | TaiSHi | angryuser: If I set the allowed codecs to 'all' or 'g729' I dont see any error message in console |
| 15:26.44 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 15:26.50 | TaiSHi | If I set to disallow all, allow ulaw (i.e.) |
| 15:26.55 | TaiSHi | I get 'No compatible codecs found' |
| 15:27.10 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 15:27.41 | TaiSHi | Same question goes to everyone, specially [TK]D-Fender |
| 15:27.44 | TaiSHi | (Teras Kasi?) |
| 15:28.29 | [TK]D-Fender | TaiSHi: So... where's the QUESTION? |
| 15:28.39 | TaiSHi | What could be happening ? :P |
| 15:30.22 | angryuser | kill the geek |
| 15:30.24 | angryuser | ;) |
| 15:30.35 | [TK]D-Fender | TaiSHi: Pretty obvious that the device doesn't speak only ULAW |
| 15:30.42 | mercestes | Gah, is there a polycom update for the new timezones? |
| 15:31.00 | [TK]D-Fender | mercestes: 2.1.0 covers it |
| 15:32.25 | TaiSHi | [TK]D-Fender: My ITSP speaks g729 |
| 15:32.33 | TaiSHi | But if I allow g729 in any way |
| 15:32.43 | TaiSHi | I will just get 'busy' and no error message :( |
| 15:33.15 | [TK]D-Fender | TaiSHi: you just said that if you se to G.729 that you DON'T get error messages |
| 15:33.32 | [TK]D-Fender | TaiSHi: Check your G.729 licenses, and your phones. |
| 15:34.24 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:35.20 | TaiSHi | [TK]D-Fender: No phones on this side, voice menu |
| 15:35.25 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:35.28 | TaiSHi | Open g729 |
| 15:35.42 | [TK]D-Fender | TaiSHi: ... |
| 15:35.44 | [TK]D-Fender | ~wglwat |
| 15:35.47 | jbot | wglwat is, like, well, good luck with all that |
| 15:39.04 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 15:39.37 | TaiSHi | omg.... |
| 15:39.42 | TaiSHi | I just got pwned >:< |
| 15:39.45 | angryuser | what does numbers men in 'show translation' table? (from one codec to another) |
| 15:40.00 | angryuser | *mean |
| 15:40.10 | TaiSHi | miliseconds, charge ? |
| 15:40.57 | TaiSHi | [TK]D-Fender: Bad attitude =( |
| 15:41.47 | [TK]D-Fender | TaiSHi: Sorry, just means that I have no idea how to assist debugging it, nor can I confirm compatability, and you've got low odds on anybody else either. |
| 15:42.10 | CrashSys | So sayeth the fender, so sayeth the flock... |
| 15:42.15 | TaiSHi | Well, I dont express myself very well |
| 15:42.30 | mercestes | you expressed yourself pretty well with "open g729". |
| 15:42.35 | CrashSys | I think i'm going to go home sick... I am DST '07 incompatible and hence am waiting for an upgrade... |
| 15:43.15 | TaiSHi | mercestes: ? |
| 15:45.41 | mercestes | TaiSHi: we don't do open g729 here. We pay the $10 with pride. |
| 15:46.24 | [TK]D-Fender | CrashSys: So go upgrade! |
| 15:46.27 | mercestes | ....actually....most of us don't even use g729. |
| 15:46.30 | [TK]D-Fender | Its what I'm doing now... |
| 15:46.57 | mercestes | extranet.polycom.com is *SLOW* ... =/ Polycom is the pennacle of crappy customer service. |
| 15:47.23 | CrashSys | This outta be interesting... running sip 2.1.0 on an IP301 |
| 15:52.36 | mercestes | Someone was complaining about the 301, too many buttons to get to "transfer", screen wasn't big enough, resolution was too bad, speakerphone sounded like ass. It's like...what do you expect when you order the cheapest phone available? gah! |
| 15:52.48 | [TK]D-Fender | CrashSys: Don't see how it'd be "interesting". There were no noticable changes for that platform... |
| 15:53.49 | [TK]D-Fender | mercestes: Its a SPEAKER, not SPEAKERPHONNE so I wouldn't expect miracles, res = duh, 1 "more" to get to Transfer IIRC, so big deal. |
| 15:54.05 | mercestes | Yea, but you know how end users are. |
| 15:54.08 | [TK]D-Fender | mercestes: Does its job just fine. If they have that many calls, get them a higher model :) |
| 15:54.29 | [TK]D-Fender | mercestes: Yes, complete tight-wads who whine when they get their money's worth.... |
| 15:54.49 | [TK]D-Fender | Warning to all : Prepare to reap what you have sown! |
| 15:55.40 | mercestes | Yes! |
| 15:57.17 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 15:58.09 | Dr-Linux | Qwell |
| 15:58.11 | Dr-Linux | Qwell[] |
| 15:58.32 | *** join/#asterisk vlt (n=dm@p54B33AB7.dip0.t-ipconnect.de) |
| 16:00.11 | vlt | Hello. Why is "_0NXXX." stronger than "_0[79]00XXX." when dialling 070055555? |
| 16:01.57 | CrashSys | By interesting I mean I wonder how long the load time will be... |
| 16:02.26 | [TK]D-Fender | CrashSys: Boot time is more or less unchanged per platform, full flash <>5 mins |
| 16:02.49 | [TK]D-Fender | CrashSys: Falls firmly under the category of "whatever" for me... |
| 16:02.56 | CrashSys | Seems like every sip upgrade takes a little longer to load the image when booting... not that i've ever really timed it... just by looking... |
| 16:03.28 | CrashSys | it's not an issue |
| 16:04.11 | CrashSys | Just one of those things... you install upgrades and you notice little differences... |
| 16:04.44 | vlt | When omitting the "[79]" part and defining two extensions "_0700XXX." and "_0900XXX." it works ... |
| 16:04.59 | angryuser | vlt:i woul prefer _0[79]XXXXXXX,1, n mean any number [79] is 7 or 9 |
| 16:05.54 | angryuser | vlt: try to not use . dangerous |
| 16:06.19 | vlt | angryuser: But I don't know how long the number will be (living in .de) |
| 16:06.50 | angryuser | vlt: bad for you;) |
| 16:07.37 | *** join/#asterisk stack_ (n=stack@63.239.190.203) |
| 16:07.40 | vlt | angryuser: "N" means [2-9], covering [79], too ... |
| 16:08.13 | stack_ | I made the changes to my sip.cfg for my Polycom phones for DST, but they don't seem to recognize it... anyone else get this working? |
| 16:08.21 | angryuser | vlt: shouls work normally try [1-9] |
| 16:08.33 | CrashSys | Sip 2.1.0 |
| 16:08.34 | CrashSys | :) |
| 16:08.43 | angryuser | vlt:i mean [2-9] |
| 16:08.46 | vlt | angryuser: will try [2-9] ... |
| 16:08.49 | vlt | <PROTECTED> |
| 16:11.14 | vlt | angryuser: hmmm strange, "_0[2-9]XXX." beats "_0[79]XXX." when dialling "070055555" ... |
| 16:11.28 | vlt | angryuser: sorry ... wrong |
| 16:11.35 | vlt | angryuser: hmmm strange, "_0[2-9]XXX." beats "_0[79]00XXX." when dialling "070055555" ... |
| 16:11.43 | vlt | angryuser: ^^^ right |
| 16:14.18 | angryuser | vlt:beats? you put them together in dialplan? |
| 16:23.27 | vlt | angryuser: Errrm, yes, there's "_0[2-9]XXX." and "_0[79]00XXX.". The first covering all numbers beginning with 0[2-9] and at least 4 more digits. The second patterns defines an exception for numbers beginning with 0[79]00 and at least 3 more digits. Is this wrong? |
| 16:25.52 | angryuser | vlt: no it is not wrong if you need to define different action to each group |
| 16:27.21 | stack_ | does anyone have their DST working on their Polycom phones? |
| 16:28.29 | vlt | angryuser: Yes, I need to because 0[79]00. is not covered by my phone flat rate. |
| 16:29.15 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 16:30.42 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 16:30.57 | mercestes | Can someone get me the 2.1.0 Sip image and teh 3.2.2 bootrom please? |
| 16:31.26 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:31.54 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:33.02 | tuan_modulis | is there a way to hide user input (from a read function) in the CLI? |
| 16:33.46 | Qwell[] | _VoicePulse: Way to shamelessly advertise |
| 16:36.18 | *** join/#asterisk eald (n=eald@189.157.105.23) |
| 16:41.18 | wunderkin | TelcomJoshVoxMart |
| 16:41.23 | *** part/#asterisk wasabi_ (n=jhaltom@ubuntu/member/wasabi) |
| 16:41.45 | [TK]D-Fender | Qwell[]: where is this? I don't see any funny quit messages or chatter... |
| 16:41.55 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-36938dc84960c636) |
| 16:41.56 | wunderkin | just his nickname i guess.. :D |
| 16:42.04 | Qwell[] | about an hour ago |
| 16:42.12 | wunderkin | oh.. brain lag |
| 16:44.08 | mercestes | Heh, nice. I was just in CA too. |
| 16:44.22 | mercestes | I think abusing IRC for free advertising is automatic bad press. |
| 16:44.42 | Qwell[] | mercestes: pretty much |
| 16:44.50 | *** join/#asterisk xNinja (n=Bigheart@80.184.14.76) |
| 16:44.52 | xNinja | hello |
| 16:45.09 | TaiSHi | Qwell[]: I would like to make a consult to you regarding my problem |
| 16:45.20 | TaiSHi | Since guys said something like |
| 16:45.23 | Qwell[] | TaiSHi: no can do |
| 16:45.23 | TaiSHi | ~wglwt |
| 16:45.30 | Qwell[] | ~wglwat |
| 16:45.31 | jbot | methinks wglwat is well, good luck with all that |
| 16:45.35 | TaiSHi | Ok, that |
| 16:45.38 | TaiSHi | >.< |
| 16:45.47 | wunderkin | wiggilwat? |
| 16:45.48 | xNinja | hello...i can find help here about .mo slax modules ? |
| 16:45.53 | mercestes | TaiSHi: Why not just pay for the g729 licenses and become compliant? |
| 16:46.22 | [TK]D-Fender | xNinja: Nope, try #slax |
| 16:46.27 | Qwell[] | TaiSHi: nobody is gonna help you infringe on somebodys patent |
| 16:46.52 | mercestes | For anything less than $100 an hour. >.> |
| 16:46.55 | xNinja | #slax little users and no one answering |
| 16:47.11 | Qwell[] | xNinja: well, this is definitely the wrong channel |
| 16:47.15 | bkruse | that shoudl tell you something about slackware ;P |
| 16:47.20 | [TK]D-Fender | xNinja: Sorry, but this is not "backup distro help channel" |
| 16:47.27 | Qwell[] | bkruse: slackware is troll-proof |
| 16:47.35 | bkruse | Qwell[]: exactly........ :P |
| 16:47.42 | Qwell[] | bkruse: I mean, you can't troll on it :p |
| 16:48.05 | Qwell[] | it's got street cred |
| 16:48.08 | xNinja | lol |
| 16:48.23 | bkruse | Qwell[]: it does, i use slackware at home |
| 16:48.55 | [TK]D-Fender | bkruse: Ditto, though I'm swithcing the CentOS when I "get around To It" |
| 16:48.56 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 16:49.00 | tutt9876 | hi |
| 16:49.06 | xNinja | ok thanks |
| 16:49.08 | xNinja | salam |
| 16:49.10 | *** part/#asterisk xNinja (n=Bigheart@80.184.14.76) |
| 16:49.47 | Qwell[] | salam? |
| 16:49.47 | wunderkin | balogna |
| 16:49.47 | TaiSHi | Qwell[]: It is not my idea to infringe patents, yet, Open G729 is ... open |
| 16:49.48 | bkruse | [TK]D-Fender: ya, im prolly switching to something else, but i love it |
| 16:49.48 | Qwell[] | TaiSHi: Open != Free |
| 16:49.48 | [TK]D-Fender | xninga : Bologna ;) |
| 16:49.48 | TaiSHi | Also, I'm not using it for commercial purposes |
| 16:49.49 | TaiSHi | comercial * |
| 16:49.50 | Qwell[] | TaiSHi: patents are patents - commercial or not |
| 16:49.55 | [TK]D-Fender | wunderkin: Beat me to it! |
| 16:50.19 | [TK]D-Fender | wunderkin: Might not have if not for time spent correcting typos :) |
| 16:50.25 | mercestes | TaiSHi: What do we look like, the morality cops? We don't care what yoru doing with OpenG729...we just can't help you perpetrate your infringment. |
| 16:50.27 | Qwell[] | TaiSHi: You aren't going to get support here for that - end of story. |
| 16:50.31 | [TK]D-Fender | wunderkin: ...... *TMI* |
| 16:50.53 | TaiSHi | Qwell[]: Roger, about G729, it's 10$ per channel |
| 16:51.02 | Qwell[] | that's EXTREMELY cheap |
| 16:51.05 | TaiSHi | I know |
| 16:51.09 | TaiSHi | (not for Argentina tho) |
| 16:51.09 | Qwell[] | So what's the problem? |
| 16:51.17 | Qwell[] | then don't use g729... |
| 16:51.26 | mercestes | I'm pretty sure they'll take virgins. |
| 16:51.27 | gmaxwell | Use speex instead... |
| 16:51.28 | Qwell[] | mercestes: gladly |
| 16:51.31 | mercestes | and if they dont' I can facilitate an exchange rate for you. |
| 16:51.35 | Qwell[] | mercestes: :P |
| 16:51.38 | mercestes | :D |
| 16:51.39 | TaiSHi | My provider uses only G729 |
| 16:51.48 | mercestes | TaiSHi: Then get a diff. provider. |
| 16:51.50 | Qwell[] | TaiSHi: Then get a different provider, or pay the fee |
| 16:54.13 | TaiSHi | :P Tell that to my boss |
| 16:54.24 | TaiSHi | Bah, screw him u_U |
| 16:54.24 | Qwell[] | bring him on here |
| 16:54.36 | defend | lol |
| 16:54.39 | TaiSHi | Will do if he says another word regarding the problem |
| 16:54.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:54.45 | defend | ya let qwell at him |
| 16:54.54 | TaiSHi | Qwell[]: He's big |
| 16:54.55 | TaiSHi | trust me |
| 16:55.02 | Qwell[] | I don't care. |
| 16:55.20 | bkruse | TaiSHi: buy a transcoder card |
| 16:55.23 | [TK]D-Fender | TaiSHi: Big = slow, slow = dead. |
| 16:55.31 | vlt | Hello. Is there an option for cmd Playback() to mix two audio files in realtime? |
| 16:55.39 | bkruse | lol |
| 16:55.42 | Qwell[] | vlt: nope... |
| 16:55.47 | wunderkin | big / little = dead |
| 16:55.51 | bkruse | sorry i just found that pointless |
| 16:56.04 | vlt | Qwell[]: Thanks. |
| 16:56.12 | [TK]D-Fender | wunderkin: Enough of you and your asphyxiation fetish ok?! |
| 16:56.20 | wunderkin | heh |
| 16:56.31 | mercestes | ... |
| 16:56.31 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 16:56.33 | TaiSHi | bkruse: ? |
| 16:56.34 | diclophis-work | hello all |
| 16:56.37 | mercestes | asphyxiation fetish? Yay |
| 16:56.48 | diclophis-work | i have a question about picking up and hanging up really fast |
| 16:56.55 | TaiSHi | Imagine him falling on you Qwell[] |
| 16:56.59 | TaiSHi | Like |
| 16:57.01 | TaiSHi | zomg the Lphant |
| 16:57.08 | TaiSHi | (power of p2p u_U) |
| 16:57.33 | [TK]D-Fender | diclophis-work: It's called "pulse dialing" :) |
| 16:57.41 | bkruse | TaiSHi: buy a transcoder card, http://www.digium.com/en/products/hardware/voicecards.php |
| 16:57.44 | diclophis-work | particularly with cell phone because there seems to be some sort of lag in general |
| 16:57.52 | diclophis-work | [TK]D-Fender: hah, true |
| 16:58.00 | tutt9876 | got a problem to match an extension beginning a letter: I have exten => [a-zA-Z],2, in extensions.conf but it's never reached |
| 16:58.17 | tutt9876 | beginning with a letter |
| 16:58.18 | diclophis-work | the wierd thing is that it seems to send the call into some sort of wierd state in the dialplan |
| 16:58.44 | diclophis-work | i am also dialing with a "G"oto option |
| 16:59.03 | tutt9876 | any indea how to match a begining with a letter extension? |
| 16:59.41 | Qwell[] | tutt9876: prepend it with a _ |
| 16:59.43 | angryuser | goodbye everybody |
| 16:59.46 | diclophis-work | tutt9876: well most phones cant dial alpa extensions? |
| 17:00.00 | sergee | can i read somewhere about "int realtimeregs = ast_check_realtime("sipregs");" - what is this? new instance of RT ? |
| 17:00.05 | tutt9876 | yes I use softphone |
| 17:01.20 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:01.28 | tutt9876 | how to match this extension: Iambeginningwithletter ? |
| 17:02.25 | [TK]D-Fender | tutt9876: Qwell[] just gave you the answer. Pay attention. |
| 17:02.26 | Corydon-w | tutt9876: you're missing the leading underscore |
| 17:02.54 | tutt9876 | ok I just understood my mistake by the way : thanks |
| 17:03.02 | putzz | what would cause incoming did to not go thru and show: handle_request_invite: Failed to authenticate user. when I can make outgoing calls and receive calls on other dids(diff provider) |
| 17:05.31 | *** join/#asterisk jsolares (n=jsolares@200.12.34.80) |
| 17:05.51 | jsolares | Hi, does anyone know if IAX2 is able to transmit audio before an answer is sent? |
| 17:06.23 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:06.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:07.14 | zoa | jsolares: it is |
| 17:07.46 | mercestes | Gah, gotta fix the Cisco 7940's too. *sighs* Gonna be a long day. |
| 17:09.07 | jsolares | i wonder if chan_unicall can as well :S |
| 17:09.10 | jsolares | meh i'm starting to get annoyed by this project |
| 17:09.51 | jsolares | already had to modify asterisk sources to have number not here and subscriber error messages from R2 sent back to ISDN on another machine |
| 17:10.02 | CrashSys | Anyone familiar with Cisco IAD2432's? Seems like it might be dropping calls and just wondering if anyone else has ever worked with one of these... |
| 17:10.42 | jsolares | thanks zoa :) |
| 17:12.51 | CrashSys | Maybe they will do away with DST.. |
| 17:21.10 | Corydon-w | CrashSys: not a chance |
| 17:21.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:22.22 | CrashSys | I can dream I guess... |
| 17:22.26 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:37.35 | *** join/#asterisk doolph (n=doolph@200.105.35.219) |
| 17:38.54 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:39.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:41.40 | bkruse | #asterisk is so quite |
| 17:41.54 | mut | and quiet |
| 17:41.55 | bkruse | quiet* |
| 17:41.59 | bkruse | lol, ty ;] |
| 17:44.45 | [TK]D-Fender | SHHHH!!!! You'll wake the crickets! |
| 17:45.17 | Cyon | *creak* *creak* |
| 17:45.22 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 17:45.56 | doolph | anybody know how to control ingress bandwidth ? |
| 17:46.00 | Corydon-w | Everyone is off fixing the friggin' time change |
| 17:46.35 | Corydon-w | Can I bill Congress for the time it takes me to fix everything? |
| 17:46.44 | Cyon | doolph: If you run cisco, jump into #cisco and you can do it with policy-maps, policers, etc. |
| 17:47.39 | CunningPike | Corydon-w: The energy savings will pay for it ;) |
| 17:47.55 | CunningPike | Corydon-w: Only problem is, the ROI is 7,543 years |
| 17:48.18 | Corydon-w | Problem is, Congress will change it again long before then |
| 17:49.14 | Corydon-w | Maybe if everybody sent a bill for the cost to the Congressman who proposed this, he'll think before changing it again |
| 17:49.18 | Cyon | It's a downhill spiral until we convert the entire world to the same time; and people just get up at different times; remove the relationship between time and daylight, and who cares. |
| 17:49.58 | Corydon-w | Cyon: I would tend to agree. Just set all clocks to UTC and we'll deal with the localization locally |
| 17:50.04 | doolph | Cyon i am using linux :D |
| 17:50.25 | doolph | my lan shaping is working fine, but the server cannot be shaped |
| 17:50.29 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:50.34 | doolph | otherwise I'll need another server |
| 17:50.46 | caio1982 | is the ToHost parameter cached for a given sip peer? I've found that only a asterisk restart restore the default behavior after changing it to a random hostname... |
| 17:50.57 | *** join/#asterisk adde (n=adde@tn-84-218-53-190.dsl.bredbandsbolaget.se) |
| 17:51.16 | caio1982 | (i mean, host=dynamic is ok but further changes cant be undone until a restart) |
| 17:51.28 | CunningPike | Corydon-w: Exactly - I can't think of any other way of doing it. But Microsoft obviously did...... |
| 17:52.11 | Qwell[] | CunningPike: last I checked, it took them 3-4 updates to get it right in Vista |
| 17:52.15 | Corydon-w | CunningPike: why does anybody care if banking hours are different locally? |
| 17:52.44 | CunningPike | Qwell[]: It's still fscked |
| 17:52.49 | Idle | hmmm, is there a trick to forcing a cisco SIP phone to download the latest firmware? mines not even querying OS79XX.TXT |
| 17:52.52 | Qwell[] | doesn't surprise me |
| 17:52.53 | CunningPike | Corydon-w: Not me |
| 17:52.58 | Qwell[] | Idle: new ones don't |
| 17:52.59 | Corydon-w | It only really matters in the US when the Eastern seaboard closes |
| 17:53.01 | Qwell[] | Idle: sip? |
| 17:53.06 | Idle | yea |
| 17:53.15 | Qwell[] | hmm, nevermind, I don't have my config files here |
| 17:53.18 | Idle | hoping the latest has DST, and whatnot |
| 17:53.30 | Qwell[] | I think in the SEP<mac>.cnf.xml you just tell it the firmware filename |
| 17:53.33 | Idle | according to cisco's site, OS79XX is supposed to download |
| 17:53.44 | Qwell[] | new firmwares read the SEP<mac> file before all others |
| 17:53.49 | Qwell[] | *NOT* the SIP<mac> |
| 17:54.07 | Idle | it doesn't even query for that |
| 17:54.18 | Idle | oh, wait, there it is |
| 17:54.21 | Qwell[] | :D |
| 17:54.29 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 17:54.29 | Qwell[] | find a copy on the wiki |
| 17:54.31 | Qwell[] | ~wikis |
| 17:54.33 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 17:55.29 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:55.43 | Idle | aight |
| 17:55.48 | Idle | lets see |
| 17:55.59 | Idle | son of a |
| 17:56.03 | Idle | still loading the OLD one |
| 17:56.10 | Idle | pile of crap criscp |
| 17:56.26 | adde | i am trying to get a simple menu working on my asterisk.... but i get the soundfile played but i cant make any choices it waits for a while and then hungs up.... Could some kind soul just take a look at theese line: http://pastebin.ca/392116 |
| 17:57.24 | *** join/#asterisk stony (n=stony@p5B153425.dip0.t-ipconnect.de) |
| 17:57.38 | stony | <PROTECTED> |
| 17:57.39 | Idle | wrong mac :P |
| 17:57.50 | Idle | there we go |
| 17:57.57 | stony | does anyone know a mirror of the freepbx doc ? aussievoip.com is dead ... |
| 17:58.04 | Idle | now, what do I do once I brick it? |
| 17:58.15 | clyrrad | Has anyone here set the daylight saving time rule on a Sipura device? I have added the rule but the time does not change. The rule I am using is GMT-05:00 and start=3/2/7/02:0:0;end=11/1/7/02:0:0;save=1 - our timezone is EST so GMT-5, but due to DST we are GMT-4 now. Do I need to update GMT-4 manually to get the time right? |
| 17:58.47 | clyrrad | I am under the impression the rule should do it automatically.... |
| 17:59.39 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:59.40 | Dr-Linux | Qwell |
| 17:59.42 | Dr-Linux | Qwell[] |
| 18:00.05 | Idle | wow.... 4 reboots |
| 18:00.08 | Idle | this cant be good |
| 18:00.41 | adde | Please could somone just take a look at my 6 lines for creating a menu... |
| 18:01.33 | Idle | huh... keeps loading, then requesting configuration, the 'resetting' |
| 18:03.15 | Idle | and now it reverted |
| 18:03.16 | *** join/#asterisk qdk_ (n=qdk@193.164.155.44) |
| 18:04.06 | SplasPood | Anyone here using Polycom phones having DST issues |
| 18:05.11 | Qwell[] | SplasPood: see asterisk-users mailing list |
| 18:07.33 | SplasPood | Qwell[]: will do |
| 18:09.05 | SplasPood | ahh.. |
| 18:09.42 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:09.56 | [TK]D-Fender | adde: I'd bet your DTMF mode isn't set right and * just isn't getting the digits |
| 18:10.20 | EmleyMoor | Is there a way to determine the reason caller ID is not presented on calls coming in over (a) Zap (b) IAX2 (c) SIP? |
| 18:12.59 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:13.40 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 18:14.29 | clyrrad | anyone know about my DST question? |
| 18:14.31 | *** join/#asterisk |kahless| (n=kvirc@i577ACEC2.versanet.de) |
| 18:14.37 | |kahless| | hello all |
| 18:15.18 | [TK]D-Fender | EmleyMoor: a: not provided by telco, not supported by your card, or * has no idea about your regions CID signalling method, or you screwed up its config |
| 18:16.14 | EmleyMoor | [TK]D-Fender: That's not what I meant |
| 18:17.00 | EmleyMoor | I meant when a call comes in, how to tell if the caller withheld the number, it was unavailable or it is not possible due to it being international (if, of course, it's not presented) |
| 18:18.41 | EmleyMoor | I know I get the name "RINGBACK" on ringbacks, but not sure how to tell the rest |
| 18:19.03 | *** join/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 18:20.22 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 18:22.02 | SplasPood | Qwell[]: hrm.. interesting.. even tho I'm running 2.1.0 on my phone here, and its using the 2.1 config with the date changes already done it doesn't show the DST change. its still 1 hr behind ... I'm gonna see what my dhcp server is offering up for gmt offset, etc |
| 18:22.10 | [TK]D-Fender | EmleyMoor: Block a call, call in and SEE |
| 18:22.33 | [TK]D-Fender | SplasPood: Make sure your sip.cfg is OVERRIDING DHCP |
| 18:22.39 | EmleyMoor | Tried that... can't tell how to tell the difference |
| 18:22.46 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-219-226.dsl.irvnca.pacbell.net) |
| 18:23.08 | bsd_tech | ok we got asterisk - asterisk addons working on bsd with the new zaptel |
| 18:23.10 | bsd_tech | yes |
| 18:23.24 | [TK]D-Fender | EmleyMoor: Then hate your telco.... |
| 18:23.35 | EmleyMoor | All I get is empty in the CALLERID(all) |
| 18:23.42 | EmleyMoor | (on a withheld) |
| 18:23.46 | [TK]D-Fender | EmleyMoor: Then take that as you will. |
| 18:24.01 | EmleyMoor | No idea how, if at all, I can determine the others |
| 18:24.06 | *** join/#asterisk redox- (i=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com) |
| 18:24.16 | [TK]D-Fender | EmleyMoor: Again, try & see |
| 18:24.18 | EmleyMoor | At the moment, I treat them all as withheld and prevent the phones ringing |
| 18:24.28 | EmleyMoor | No idea how I can even try |
| 18:24.41 | tuan_modulis | gah, there's not much on security for voip |
| 18:25.14 | EmleyMoor | I get caller ID from Belgium over IAX - I know that because it happened the other day |
| 18:25.41 | redox- | What does this line in the log file mean: "acl.c ##### Testing 172.20.10.5 with 172.20.0.0" ? |
| 18:26.00 | *** join/#asterisk alvariux (n=alvaro@189.158.154.44) |
| 18:26.04 | alvariux | hello |
| 18:26.33 | zoa | EmleyMoor: it doesnt have to be there |
| 18:26.37 | zoa | it could be missing |
| 18:26.41 | alvariux | is there a channel about iaxclinet ? i tried #iaxclient but is empty |
| 18:26.57 | EmleyMoor | zoa: Can you please explain that comment? |
| 18:27.06 | alvariux | i need help on building iaxclient in windows using cygwin |
| 18:27.16 | alvariux | any advice ? |
| 18:28.30 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 18:28.51 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 18:29.05 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 18:29.05 | tuan_modulis | anyone know anything about encrypting parameter values passed to AGI? |
| 18:29.32 | zoa | i mean there are different reasons why it could be missing |
| 18:29.33 | tuan_modulis | i mean, an agi script |
| 18:29.42 | Assid | err.. anyone know the changes to do in sip.cfg for polycoms for the new dst ? |
| 18:29.44 | alvariux | someone have build iaxclient with cygwin? |
| 18:29.46 | zoa | simboxes, carrier not passing it, hidden from the sender etc |
| 18:30.16 | EmleyMoor | zoa: I want to know the reason it is missing - caller ID boxes present the reason |
| 18:30.31 | zoa | no they dont |
| 18:30.59 | EmleyMoor | Yes they do - they say "WITHHELD", "UNAVAILABLE" or "INTERNATIONAL" |
| 18:31.05 | Braxus | clyrrad; try this rule... this is the one I did and it seemed to work for DST: start=3/11/0:2:0:0;end=11/1/7/2:0:0;save=1 |
| 18:31.09 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 18:31.32 | zoa | that must be something for normal analogue lines then |
| 18:31.43 | zoa | on isdn its not possible |
| 18:32.24 | Braxus | I've gotten DST to work on polycoms. Had to specify a manual date under sip.cfg. |
| 18:33.27 | Assid | Braxus: care to give me that line from the cfg |
| 18:33.35 | Braxus | sure hold on |
| 18:33.53 | [TK]D-Fender | Braxus: pre SIP 2.1.0 you can set it for the first week of March and think about the long term fix for next year. the latest firmware lets do this precisely |
| 18:34.44 | mcab | http://tinyurl.com/ynokqq <-- polycom tech-bulletin for the DST changes |
| 18:34.46 | |kahless| | hi :) which "asterisk/asterisk-bundle-whatever-iso/other" version do you recommend a german fritz card (A1 ISDN [Fritz] (rev 02)) isdn newbie? |
| 18:34.54 | Braxus | DST starts on 2nd week of march? |
| 18:35.13 | Assid | i need to update sip there then |
| 18:35.23 | Assid | [TK]D-Fender: update sip and bootrom? |
| 18:35.35 | [TK]D-Fender | Assid: just SIP |
| 18:35.39 | mcab | Assid: no, just a config update |
| 18:35.47 | [TK]D-Fender | Assid: BootROM has nothing to do with anything really. |
| 18:35.56 | Assid | nah just thinking if i should do both |
| 18:36.28 | mcab | 2.1.0 comes with the DST changes in the configs, but prior SIP software will handle it fine with the right config tweaks |
| 18:36.28 | Braxus | http://pastebin.ca/392175 |
| 18:36.34 | Braxus | ah |
| 18:36.49 | toombaloomba | lol thats funny everyone is doing what im doing |
| 18:36.49 | toombaloomba | goddamn DST :( |
| 18:36.55 | Braxus | when I did it, wasn't runing 2.1.0 just yet. heh |
| 18:37.35 | Braxus | DST started yesterday... at least in the US. |
| 18:37.54 | [TK]D-Fender | mcab: Not really... previous versions couldn't handle the start date on the SECOND week of a month... |
| 18:38.41 | Braxus | on linksys/sipura phones, I couldn't set DST to start on the 2nd sunday of march. only 1st or last weeks. |
| 18:40.06 | Assid | Braxus: you in EST ? |
| 18:40.16 | Braxus | hey is there a newer bootrom for polycom phones above 3.2.2? Read somewhere there is but couldn't find anything regarding it |
| 18:40.22 | Braxus | Assid: PST |
| 18:40.27 | Qwell[] | PDT |
| 18:40.41 | Braxus | okay PDT :P |
| 18:40.53 | Qwell[] | or better, PST8PDT |
| 18:42.07 | Braxus | running 3.2.2 bootrom and SIP 2.1.0 now. Here there is a 3.2.3b for the bootrom part. |
| 18:43.53 | mcab | [TK]D-Fender: really? don't see anything about that in the release notes, and the tech-bulletin doesn't say that you need ot upgrade to 2.1.0... |
| 18:44.24 | *** join/#asterisk redox- (i=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com) |
| 18:45.29 | [TK]D-Fender | mcab: I THINK 2.0.3B might have it... |
| 18:46.01 | [TK]D-Fender | mcab : Confirmed... thats what My addendum guide says... |
| 18:46.59 | EmleyMoor | Hmmm... whether I SetCallerPres(unavailable) or (prohib), it withholds as far as BT is concerned |
| 18:47.16 | [TK]D-Fender | mcab: s'ok, I research so nobody here HAS to ;) |
| 18:47.18 | CunningPike | fwiw, we're running 1.6.7 and made the changes in the Polycom bulletin and they worked perfectly |
| 18:48.33 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:49.39 | Braxus | curious, any of you guys know of how to auto-provision the personal directories in sipura/linksys phones? |
| 18:51.06 | *** part/#asterisk |kahless| (n=kvirc@i577ACEC2.versanet.de) |
| 18:51.20 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 18:53.41 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 18:54.51 | Assid | brb |
| 18:55.19 | *** join/#asterisk Arth_72 (n=Arturo@88-149-232-233.f5.ngi.it) |
| 18:55.31 | wunderkin | well we still have reboots but the key problem may be gone |
| 18:56.02 | wunderkin | it seems to me very odd that the 501s are fine but the 430s have problems :P they rebooted while on a power brick too.... |
| 18:56.18 | Braxus | 430s have problems? |
| 18:56.25 | Braxus | was considering getting a couple 430s here |
| 18:56.26 | wunderkin | just me i think! |
| 18:56.42 | Arth_72 | hi |
| 18:56.47 | Braxus | how are the speakerphones on the 430s compared to 501s? |
| 18:56.48 | wunderkin | i like the 501s better though |
| 18:57.02 | wunderkin | probably the same, i dont really use it |
| 18:57.32 | [TK]D-Fender | Braxus: Do you already run PoE? |
| 18:57.41 | Braxus | nope |
| 18:57.54 | wunderkin | if the key problem is fixed, that means that the cpu can not handle 60 sec registrations.. which is way whack... |
| 18:59.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:00.02 | toombaloomba | newest cisco 79x0 SIP firmware is fixed for new DST right? |
| 19:00.16 | [hC] | so, im going to assume that the daylight savings crap is going to have to be updated in phone config files for cisco and polycom? Im kind of annoyed that it doesnt just take it from the NTP server |
| 19:00.21 | [hC] | but i guess it has to know.. |
| 19:00.34 | Qwell[] | [hC]: ntp uses utc |
| 19:00.41 | Qwell[] | quite rightly |
| 19:00.41 | [hC] | yeah. |
| 19:00.48 | [hC] | just realized that as i typed it |
| 19:00.50 | [TK]D-Fender | [hC]: DST comes from the LOCAL config, so yup, its topic'o'the'day :) |
| 19:01.02 | [hC] | haha surprise surprise. |
| 19:01.42 | [hC] | anyone posted polycom/cisco config file changes yet so i dont have to think today? :P |
| 19:02.07 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 19:07.54 | diclophis-work | should you always have a "Congestion" command after a "Dial" command? |
| 19:07.58 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 19:08.22 | Qwell[] | no, there are more reasons a call could "fail" than just congestion |
| 19:08.59 | diclophis-work | so i need some logic after the dial command to switch based on HANGUPCAUSE and DIALSTATUS? |
| 19:12.18 | diclophis-work | Goto(x-${DIALSTATUS},1) is a valid goto? |
| 19:15.04 | Qwell[] | sure |
| 19:16.22 | EmleyMoor | 849 priorities in my dialplan now |
| 19:17.12 | *** join/#asterisk Assid (n=assid@59.183.29.37) |
| 19:17.56 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
| 19:17.57 | diclophis-work | ok, so how come the Dial command doesnt return immediatly after i dial(zap/g1/cellnumber), and then on my cell i press the answer button, then immediatly hangup |
| 19:18.12 | Qwell[] | because cellphones are slow |
| 19:18.13 | diclophis-work | it seems to get stuck at the Dial command for about 30 seconds |
| 19:18.18 | Qwell[] | very slow |
| 19:18.28 | file | Qwell[]: you're slow! |
| 19:18.29 | diclophis-work | so thats normal operating behaviour |
| 19:18.39 | Qwell[] | and it's compounded if you're using an x100p, or something without hangup detection |
| 19:18.43 | diclophis-work | it does come back as "CHANUNAVAIL" |
| 19:18.54 | EmleyMoor | I can't work out how I'm ever going to work out the difference between "WITHHELD", "UNAVAILABLE" and "INTERNATIONAL" |
| 19:18.56 | diclophis-work | mm its a t1 card |
| 19:19.21 | Corydon-w | You're more or less exploiting the legacy behavior of the PSTN |
| 19:20.00 | mmartinn | Is there a way to secure FastAGI stuff? |
| 19:20.21 | diclophis-work | mmartinn: only run it on an internal network? |
| 19:20.25 | Arth_72 | any1 can give me some hint on how can i make my asterisk detect incoming fax ? i put exten => fax ... in my dial plan (after doing an Answer) .. but it's not working ... |
| 19:20.30 | mmartinn | diclophis-work: so basically firewall :) |
| 19:20.31 | Corydon-w | mmartinn: run a VPN |
| 19:20.37 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 19:21.16 | mmartinn | I suppose I could connect to localhost with an SSH tunnel |
| 19:21.33 | mmartinn | hrm :( |
| 19:21.48 | Corydon-w | mmartinn: install OpenVPN |
| 19:22.15 | Corydon-w | Fast, cheap, effective |
| 19:22.17 | diclophis-work | Arth_72: you need to use the command FaxDetect |
| 19:22.37 | diclophis-work | NVFaxDetect to be exact |
| 19:22.43 | Arth_72 | where ? |
| 19:23.18 | diclophis-work | What I do is Answer() the channel, then set it to Ringing(), then NVFaxDetect(5) for 5 seconds, then send the call to an agi handler |
| 19:23.27 | diclophis-work | if its a fax it goes to the exten => fax extension |
| 19:23.49 | diclophis-work | what happens is a fax machine calls, and during the fake ringing plays the fax tones, and then it switches to fax mode |
| 19:23.58 | *** part/#asterisk bsd_tech (n=bsdtech@ppp-69-238-219-226.dsl.irvnca.pacbell.net) |
| 19:24.11 | diclophis-work | if its a person calling, they hear ringing, and assume its just normal line ringing, then the call is picked up and procedes as normal |
| 19:24.12 | Arth_72 | i was supposing that just doing the answer i get asterisk detecting the kind of call ... |
| 19:24.25 | diclophis-work | nope |
| 19:24.48 | Arth_72 | oh, so the wiki is not exact .. thaks a lot for the explanation ... |
| 19:24.53 | diclophis-work | you also need to have good card, and a real phone line |
| 19:25.00 | diclophis-work | or a kick ass voip service |
| 19:25.11 | Arth_72 | i have a isdn line with a hfc based chipset |
| 19:25.19 | diclophis-work | yea that should work |
| 19:25.25 | Arth_72 | nice ... |
| 19:25.30 | Arth_72 | hope so :) |
| 19:25.32 | diclophis-work | you need to hunt down the NV utilities |
| 19:25.44 | *** part/#asterisk mmartinn (n=martins@128.227.123.22) |
| 19:25.59 | Arth_72 | now i look for them ... thanks a lot again diclophis .... |
| 19:27.02 | diclophis-work | sending faxes is another story |
| 19:27.09 | diclophis-work | i still havnt got that to work |
| 19:27.46 | Arth_72 | i need only to receive them ... |
| 19:28.04 | adde | i have added a user as "friend" in sip.conf... When i try to register with softphgone the asterisk debug mode says No matching peer found.... Any ideas? ty. |
| 19:28.05 | Arth_72 | to send i use hylafax .. when the other server i have at home is up and running .. |
| 19:28.21 | Arth_72 | asterisk is just an answering machine and fax collecting when i'm out :) |
| 19:29.04 | diclophis-work | adde: do the usernames line up? and is the "host" setting in sip.conf set to dynamic, or the IP of the softphone |
| 19:31.04 | adde | name of user [100] type=user username=100 secret=test host=dynamic |
| 19:34.55 | adde | is it suppose to be type=friend? |
| 19:39.34 | *** join/#asterisk Alexo56 (n=Alexo56@201.158.1.4) |
| 19:41.28 | Alexo56 | Hi there |
| 19:44.03 | *** join/#asterisk linux4free (n=gerlach@dslb-088-064-189-001.pools.arcor-ip.net) |
| 19:44.06 | Alexo56 | I have setup a VoIP trunk in Asterisk 1.2.15 using just my username (phone no.), my provider's domain name, their server IP address and port, and my password |
| 19:44.40 | Alexo56 | registration wen fine, I can receive calls, but the outgoing call part is not working .. |
| 19:45.22 | Alexo56 | 402 ..Not Found .. that's the error |
| 19:46.34 | mercestes | Sounds like your provider's domain name doesn't resolve. Can you use the IP? |
| 19:46.42 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:47.30 | bulle | is there some simple way to find out what modules one need, for a certain setup, right now all modules load at asterisk start, and im only using sip and some basic voicemaill stuff |
| 19:48.18 | Alexo56 | I am almost sure the culprit is the port number that Asterisk ttaches in the To: field |
| 19:48.38 | mercestes | Alexo56: Why is that? |
| 19:49.23 | thekidrio | bulle, define your set up clearly and what you need from it and then module selection gets easier |
| 19:49.36 | linux4free | I am new to asterisk .. I have a very small business and want to make telephone connections with my PC. Not VoIP but with conventional ISDN lines with analog adapter an analog telephones. Can somebody give me some some pointers where start reading? Some key words?- Possibly without that VoIP-stuff. thx |
| 19:51.26 | bulle | thekidrio: my main problem seems to be to figure out what modules are needed, for the functions i am using, is there some basic doc i am missing that explains what modules are needed for what ? |
| 19:51.46 | Alexo56 | I have configured my X-Lite soft phone with my provider's info (bypassing Asterisk)using the very same info and in the To: field, X-Lite doesn't put the port number, and it works |
| 19:53.10 | Alexo56 | the port number in that field is the only thing that the X-Lite does different fron Asterisk |
| 19:54.30 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 19:54.43 | Alexo56 | Is there a way of telling Asterisk not to put the port number in the To: field? |
| 19:54.47 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:54.51 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 19:54.52 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 19:55.27 | *** join/#asterisk sathran (n=spamdiog@83.240.253.161) |
| 19:55.39 | sathran | hi everyone |
| 19:55.40 | *** join/#asterisk Assid (n=assid@59.183.56.118) |
| 19:55.42 | Assid | hrmm |
| 19:55.49 | Assid | stupid pap2 generates : SIP/2.0 489 Bad event |
| 19:56.22 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 20:00.19 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 20:00.28 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-151-204-131-147.ny325.east.verizon.net) |
| 20:00.44 | Alexo56 | Sniffing the output that the ATA my provider's gave me produces, it is the same: the ATA doesn´t put the port number in the To: field .. why Asterisk does it? |
| 20:02.05 | Assid | hey you firmware for tthat.. |
| 20:02.08 | Assid | gonna update |
| 20:02.22 | mercestes | Alexo56: Maybe you specify a portnumber in the sip.conf file, I dunno. |
| 20:02.29 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 20:02.44 | mercestes | try leaving port numbers out. I see no reason why specifying a valid port# would hurt it tho. Which provider is this? |
| 20:04.33 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 20:04.42 | *** join/#asterisk Stridernzl (n=neville@222-153-62-109.jetstream.xtra.co.nz) |
| 20:05.39 | Alexo56 | My provider is maxcom.com in Mexico .. the funny thing is if I tell Asterisk in the peer definition that my provider listens in the 5060 port (which is not true), it doesn't put any port number, but if I tell Asterisk the right port number (5085), it puts it in the To: field |
| 20:06.52 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:07.31 | mercestes | Alexo56: So put 5060 and see if it works then. |
| 20:07.53 | mercestes | 5060 would be the default American port so it does make sense that it would be omitted if the port is default, and the port included if you specify other than default. |
| 20:08.23 | mercestes | now my question is, why does you rprovider refuse your connection directly to :5085 when you specify that? |
| 20:08.37 | mercestes | Is everyone certain that they really are on port :5085? |
| 20:09.20 | Alexo56 | at registration, Asterisk does not put any port number in the To: field, but sends the request correctly to maxcom's port 5085, but in the INVITE, it does put the port number in that field, so (I presume) that makes me different at the INVITE than the registration it has |
| 20:11.02 | Alexo56 | If I use port 5060 at the INVITE, maxcom does not answer, it is listening on port 5085 ( I saw that in their ATA config) |
| 20:11.21 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 20:12.36 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:12.45 | Alexo56 | I am almost sure that is I could tell asterisk not to put that number in the To: field, maxcom.com could recognize mi INVITE request .. is there a way of doing this? |
| 20:13.37 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 20:18.25 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:22.01 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 20:24.05 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 20:24.42 | Waverly360 | Anyone here ever had trouble getting the asterisk Directory to work properly when called from an AGI script? |
| 20:25.45 | Waverly360 | Everything seems to work fine when I go into it from an IVR, but when I get to the part where I'm confirming that I want to dial a particular user, it just disconnects the call. |
| 20:26.03 | Waverly360 | I've got the proper contexts defined in the directory command. |
| 20:26.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-8-247.red.bezeqint.net) |
| 20:29.08 | JunK-Y | Waverly360: ive no problem. |
| 20:29.18 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 20:29.28 | Waverly360 | Hmm. |
| 20:29.33 | iceyp | hey guys, my asterisk starts and then dies with "Illegal instruction" at func_uri.so => (URI encode/decode dialplan functions) |
| 20:30.20 | JunK-Y | iceyp: report a bug at bugs.digium.com with backtraces. |
| 20:30.32 | iceyp | mmm |
| 20:30.44 | iceyp | Asterisk 1.4.1 |
| 20:30.56 | JunK-Y | ok, just report it. |
| 20:30.59 | iceyp | JunK-Y cant be fixed by umm removing that module? |
| 20:31.13 | JunK-Y | ya, depends of ur backtrace. |
| 20:31.31 | JunK-Y | try to remove it, but report it anyways, if it happens for ya, it happens for some1 else too. |
| 20:32.21 | iceyp | how do i do a backtrace? strace? |
| 20:32.36 | mafkees | start asterisk with -vvvvgc |
| 20:32.53 | mafkees | when it dies, it will create a file called core.XXXXX |
| 20:33.03 | mafkees | where XXXX was the pid of asterisk |
| 20:33.06 | iceyp | wikkid |
| 20:33.08 | mafkees | then you can do: |
| 20:33.17 | mafkees | gdb -c core.XXXXX /sbin/asterisk |
| 20:33.20 | mafkees | bt full |
| 20:33.48 | JunK-Y | just read the file in ur doc/ dir. |
| 20:33.53 | mafkees | lol |
| 20:33.58 | JunK-Y | doc/backtrace.txt :) |
| 20:34.15 | JunK-Y | mafkees: i saved ya tons of explanation |
| 20:34.16 | JunK-Y | heheh |
| 20:34.33 | JunK-Y | if that file isnt clear, let me know, i'll improve it. |
| 20:34.44 | *** part/#asterisk mut (n=ana@65.111.222.120) |
| 20:34.52 | *** join/#asterisk Mavvie (n=edwin@ppp42-74.lns2.syd6.internode.on.net) |
| 20:34.54 | JunK-Y | btw, when a segfault occurs, we should grab all our data itself, without any command, no? |
| 20:36.40 | mafkees | would be nice |
| 20:37.05 | mafkees | the file is clear |
| 20:37.25 | mafkees | would be convenient when there were some instructions on how to do it easily |
| 20:37.26 | JunK-Y | with grabbing the os, kernel version and stuff. |
| 20:37.38 | mafkees | I mean, sometimes a bt full gives pages and pages of output |
| 20:37.45 | mafkees | depending on what happened |
| 20:38.01 | mafkees | copy/pasting from a screen session becomes pretty annoying that way |
| 20:38.07 | JunK-Y | but if it crashes and generates everything for us, then we could just say: attach the file FOO on bugs.digium.com. |
| 20:38.12 | mafkees | I know you can automate it with a commands file |
| 20:38.24 | JunK-Y | sounds good? |
| 20:38.33 | mafkees | yeah, sounds really good |
| 20:38.36 | mafkees | can that be done ? |
| 20:38.47 | JunK-Y | ya, why not? |
| 20:38.59 | mafkees | with some shellscript you mean |
| 20:39.07 | JunK-Y | yes |
| 20:39.09 | mafkees | ah |
| 20:39.23 | mafkees | I though you were talking about some C routines in asterisk ;) |
| 20:39.40 | mafkees | was already prepared to get a nice 'oh wow, is that possible nowedays' moment |
| 20:40.01 | JunK-Y | that could be done in C too, but if * is crashing, no more routines huh :) |
| 20:40.08 | mafkees | indeed |
| 20:40.16 | mafkees | so I was really wondering if I missed something |
| 20:40.56 | mafkees | hhmm, maybe it's possible by running asterisk under systrace ;) |
| 20:41.31 | mafkees | but the shellscript to get all the info is pretty nice idea yeah |
| 20:41.38 | mafkees | makes life easier :) |
| 20:42.07 | mafkees | not that * is crashing that much. but happens enough when playing around with svn trunk and skinny channel ;) |
| 20:42.08 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:42.25 | JunK-Y | even with stable version, * is crashing sometimes. |
| 20:43.31 | mafkees | yeah |
| 20:44.03 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:45.43 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-151-205-125-171.ny325.east.verizon.net) |
| 20:46.32 | JunK-Y | mafkees: let me few days, i will see what can i do exactly with that. |
| 20:48.22 | mafkees | sure |
| 20:48.29 | mafkees | if you want me to test something .... |
| 20:48.38 | mafkees | I know how to make my box coredump |
| 20:48.55 | mafkees | I'm working on this chan_skinny stuff, and I have a line that makes it core |
| 20:51.04 | mihinomenest | so, I'm getting a 423 error when I try and register with my provider. I see that this is a result of the "expires:" header value being too short for my provider's tastes. is there a way I can set this in the config? |
| 20:51.38 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 21:07.44 | *** join/#asterisk grndslm (n=grndslm@host-64-234-4-113.nctv.com) |
| 21:08.50 | grndslm | are there any distros that focus on asterisk?? |
| 21:12.11 | *** join/#asterisk anthm (n=anthm@adsl-75-51-20-243.dsl.milwwi.sbcglobal.net) |
| 21:12.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:12.34 | doolph | asterisknow? |
| 21:12.54 | doolph | btw asterisk can be installed on any distro |
| 21:13.01 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 21:13.08 | doolph | debian, redhat |
| 21:14.38 | Assid | hrmm |
| 21:14.43 | Assid | my e61 doesnt register anymore |
| 21:17.11 | *** join/#asterisk digime (n=digime@adsl-75-11-163-104.dsl.sndg02.sbcglobal.net) |
| 21:17.35 | Assid | i dont get it |
| 21:17.37 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 21:17.41 | Assid | it sits there.. it shows it tries to register |
| 21:17.48 | Assid | err.. anyone here got their e61 working? |
| 21:18.10 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 21:19.54 | grndslm | doolph: i realize i *can* install it on any distro, but i was wondering what were the distros that *focused* on asterisk? |
| 21:20.23 | grndslm | also...what's a good modem?? i don't have one actually |
| 21:20.42 | JunK-Y | theres no linux distro focusing on * afaik |
| 21:20.47 | grndslm | a good lin compatible modem, at least |
| 21:20.54 | Assid | damn.. cant register my e61 |
| 21:21.03 | grndslm | JunK-Y: doolph above mentioned asterisknow... |
| 21:21.09 | grndslm | i know i've heard of at least one or two more |
| 21:22.35 | grndslm | ClarkConnect |
| 21:22.44 | doolph | clarkconnect is based on centos |
| 21:22.49 | grndslm | Asterisk@Home |
| 21:22.57 | doolph | its same centos also |
| 21:23.03 | doolph | its called trixbox now |
| 21:23.49 | grndslm | doolph: which one's called trixbox now? |
| 21:24.00 | doolph | asterisk@home |
| 21:24.12 | doolph | join #freepbx for more info |
| 21:25.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:25.09 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 21:30.45 | Waverly360 | Can someone give me an example of how I should be calling Directory from my agi script? "EXEC Directory voicemail-context|phone-context"? |
| 21:30.45 | Waverly360 | Maybe that's what I'm doing wrong |
| 21:32.09 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 21:32.21 | lesouvage | I'm starting with 1.4 but exten => 403,1,Setvar(TIJD_STEMPEL=${DATE}${TIMESTAMP}) doesn't seem to work anymore. I googled but I couldn't find the answer. Any suggestions? |
| 21:32.51 | AJaymn | A@H use to have a menu for the conference rooms, it would ask what conference room you want and hit #... can someone help me with that code? |
| 21:32.51 | Qwell[] | Set |
| 21:33.09 | *** join/#asterisk japerry (n=japerry@216.231.51.209) |
| 21:34.12 | lesouvage | Qwell: I tried but the variables seems to be gone. |
| 21:34.24 | lesouvage | AJaymn: something like exten => 2,3,MeetMe(,daAMsx) |
| 21:34.43 | japerry | heya folks, anyone here from the Seattle area? |
| 21:35.03 | JunK-Y | lesouvage: theres was a warning in 1.2 |
| 21:35.09 | JunK-Y | these vars are deprecated. |
| 21:35.44 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:35.47 | japerry | I'm trying to get alternatives to Verizon's local T1 service -- qwest doesn't serve redmond |
| 21:35.51 | lesouvage | JunK-Y: has there come anything in place so I can set a timestamp? |
| 21:36.22 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
| 21:36.44 | JunK-Y | yes, of course. |
| 21:36.50 | Burgwork | I have just bought a new polycom 430i and I am having issues registering it with my asterisk box |
| 21:36.52 | JunK-Y | ya had to read it when ya were using 1.2 |
| 21:36.56 | Burgwork | my old sipura phones work fine |
| 21:37.36 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 21:37.39 | lesouvage | JunK-Y: I used 1.09 and I'm in the process of moving to 1.4 so I missed all the warnings. |
| 21:37.45 | sahafeez | my stupid 501 are not updating their time right ;( |
| 21:37.51 | thekidrio | heh |
| 21:37.53 | JunK-Y | read the file in doc/ |
| 21:38.02 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 21:38.47 | *** join/#asterisk Arth_72 (n=Arturo@88-149-232-233.f5.ngi.it) |
| 21:38.53 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source PBX -=- Asterisk 1.4.1 (Mar. 2, 2007), Asterisk 1.2.16 (Mar. 2, 2007), Zaptel 1.2.15 (Mar. 2, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- DST changes for phones are being discussed on asterisk-users list |
| 21:39.33 | Assid | err.. Qwell: http://www.voip-info.org/wiki-Polycom+Phones as well.. right at the bottom |
| 21:39.45 | Arth_72 | i have problem to access to nv utilities website ... do u know if there is another site where i can get them from ? |
| 21:39.46 | Qwell[] | sure, for just polycoms |
| 21:39.47 | Assid | actually here : http://www.voip-info.org/wiki-Polycom+Phones#DaylightSavingsTimeChange |
| 21:40.26 | Qwell[] | but this affects at least polycom, cisco, sipura... |
| 21:40.26 | Assid | true |
| 21:40.26 | Assid | i went nuts tryign to configure my e61..it jsut doesnt wanna register..really odd |
| 21:40.46 | Qwell[] | I wouldn't be surprised in the slightest, to see that billing is done incorrectly for cellphones |
| 21:40.51 | Qwell[] | ie; night/weekend minutes |
| 21:41.12 | Dovid | Assid: have a look on the list archive. some one posted exat step by step on how to do it |
| 21:41.47 | Assid | Dovid: i had it working.. i dont know what i changed.. and wham doesnt wanna work |
| 21:41.54 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 21:41.55 | sahafeez | thanks for the link, i would have thought pcom would have just fix the damn thing in the last version of software they released 5 months ago. my bad |
| 21:42.25 | Dovid | i attempted once @ 4 am and gave up |
| 21:42.56 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 21:43.01 | tdi | hi all guys :) |
| 21:43.18 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-e4380ae3095753b0) |
| 21:43.20 | Qwell[] | tdi: there is probably a girl or two here also |
| 21:43.24 | Qwell[] | ahh, hell, who am I kidding? |
| 21:43.34 | tdi | Qwell[]: lol yep |
| 21:43.35 | Strom_M | i'm the closest thing you have to a girl |
| 21:43.45 | Qwell[] | Strom_M: I was gonna say it, but...yeah :P |
| 21:43.53 | Strom_M | and I get mistaken for straight ALL THE DAMN TIME |
| 21:43.55 | Dovid | hello tdi |
| 21:44.07 | Qwell[] | Strom_M: how rude of people to think that :P |
| 21:44.10 | Strom_M | hahaha |
| 21:44.11 | Corydon-w | Strom_M: I can't imagine why |
| 21:44.27 | rudholm | Strom_M: yeah, don't you hate that?!!? |
| 21:44.28 | tdi | anybody has an idea howto do recodring name in meetme, so the voice on user jopin says 'tdi joined the conference' ? |
| 21:44.30 | rudholm | gah! |
| 21:44.38 | Corydon-w | Strom_M: it's not like you're more masculine than Qwell or anything like that. :-P |
| 21:44.48 | Qwell[] | ouch |
| 21:44.49 | russellb | tdi: the 'i' option to MeetMe |
| 21:44.51 | Qwell[] | tdi: option i |
| 21:44.52 | russellb | tdi: or 'I' |
| 21:44.55 | Dovid | tdi: u want to recrod a file ? or play ? |
| 21:44.56 | Assid | im gonna try one last time.. again |
| 21:45.00 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 21:45.05 | Corydon-w | and Qwell's pretty masculine |
| 21:45.05 | Arth_72 | i have problem to access to nv utilities website ... do u know if there is another site where i can get them from ? |
| 21:45.10 | *** join/#asterisk Penggu (n=yeah@220-245-200-87.static.tpgi.com.au) |
| 21:45.22 | tdi | i can record by myself but i want some nice chick to say it |
| 21:45.25 | tdi | to everybody |
| 21:45.37 | Corydon-w | Use Cepstral, then |
| 21:45.39 | russellb | well. the person records their name ... |
| 21:45.42 | tdi | ahh we have a girl in asterisk, she lives in sounds directory |
| 21:45.48 | russellb | and then, it plays that, and then allison says "has joined the conference" |
| 21:46.04 | tdi | Corydon-w: cepstral ? |
| 21:46.11 | Corydon-w | www.cepstral.com |
| 21:46.16 | Corydon-w | text-to-speech |
| 21:46.32 | Corydon-w | It's like Festival, but better |
| 21:46.33 | tdi | aha, so the I option won't do it for me? |
| 21:46.39 | tdi | record the name and play it |
| 21:46.46 | russellb | yes, it will |
| 21:46.54 | tdi | so for me it perfect |
| 21:46.54 | russellb | just try it and you'll see what it does ..... |
| 21:47.05 | tdi | yes yes yes yes :) |
| 21:47.13 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:47.30 | tdi | reservations i can do by myself in database so this was only thing |
| 21:47.38 | tdi | thanks all guys and maybe-girls |
| 21:49.05 | thekidrio | hrmm, i just fought someone named acktdi on another channel over whats better beef or cow |
| 21:49.22 | russellb | i'm sure it was off topic there, too |
| 21:49.26 | Strom_M | hahaha |
| 21:49.32 | Strom_M | russellb++ |
| 21:50.29 | Assid | hey russellb |
| 21:50.29 | thekidrio | since when is beef ever off topic |
| 21:50.29 | thekidrio | silly to say such a thing |
| 21:50.47 | russellb | if by beef you mean, SIP, then sure |
| 21:51.00 | tdi | beef is beef |
| 21:51.11 | thekidrio | unless its cow |
| 21:51.14 | russellb | because I have some beef with SIP. |
| 21:51.25 | thekidrio | i dont want to get into the cow vs beef again tdi :P |
| 21:51.39 | sahafeez | hum, how do i remote reboot a 501? |
| 21:51.49 | tdi | ok, like i say beef is beef nothing to discuss on that |
| 21:52.02 | Penggu | hi all. i know this isn't #snom, but i couldnt dig the info - anyone here familiar with the different types of firmware updates? |
| 21:52.10 | Assid | russellb: you wanna work on that wiretrip thing later? |
| 21:52.16 | thekidrio | sahafeez: power it down eh, also it will reboot itself if you make changes |
| 21:52.37 | sahafeez | well all i have access to is the web interface and i am in san diego and it is in dc |
| 21:52.54 | thekidrio | ahh, make a change and then wait 5 mins or so |
| 21:52.57 | thekidrio | it will have rebooted |
| 21:53.09 | sahafeez | ah, pretty stupid. thanks |
| 21:53.09 | thekidrio | you could also open up a ping on its ip if ping is allowed |
| 21:53.16 | thekidrio | make changes and watch for ping time out |
| 21:53.22 | thekidrio | then when ping returns is probably back |
| 21:53.25 | thekidrio | and yes very stupid haha |
| 21:53.49 | thekidrio | i am not sure if they changed that with later phones, but i am coming to hate 501s |
| 21:53.50 | sahafeez | how hard would it have been to put a reboot button on the interface? |
| 21:54.04 | sahafeez | i have 301, 501 and 601. |
| 21:54.07 | sahafeez | not all that bad |
| 21:54.26 | JacksLivr | wasssup? |
| 21:54.42 | Qwell[] | JacksLivr: any luck with 1.4? |
| 21:54.53 | thekidrio | once it works its great, but its not very portable. i keep running into issues when i take the phone from place to place |
| 21:55.01 | thekidrio | strange issues as far as i can tell |
| 21:55.05 | JacksLivr | oh, im in scottsboro all day. on my list as soon as i get home |
| 21:55.11 | Qwell[] | ahh |
| 21:55.25 | JacksLivr | that phone will ring tonight! |
| 21:56.13 | [TK]D-Fender | thekidrio: Thats funny... one of my friend's GF is a total COW, but she's not made of BEEF :) |
| 21:56.29 | mafkees | anyone knows if it's normal that IAX2 registration from a machine running asterisk-trunk to a machine running asterisk-bristuff-1.2 is not going great all the time ? |
| 21:56.33 | thekidrio | hrmm uh oh, more fuel to the beef vs cow debate! |
| 21:56.34 | sbingner | DEAD BEEF? |
| 21:56.49 | Strom_M | DEAD BEEF CAFE |
| 21:57.05 | thekidrio | and heyas [TK]D-Fender how is your day goin? |
| 21:57.09 | Qwell[] | dead beef cafe is a facade |
| 21:57.13 | [TK]D-Fender | Strom_M: I like mine "barely dead", which is to say blue & seared :) |
| 21:57.28 | thekidrio | aka perfect |
| 21:57.41 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 21:57.59 | sbingner | 244837814094590 |
| 21:58.40 | sahafeez | dead beef. ha...my network id for novel networks back in the day. something you could spell in hex |
| 21:59.18 | Assid | [TK]D-Fender care to help me with this sip client connecting to a box? it doesnt register |
| 22:00.56 | *** join/#asterisk DaneM (n=DaneM@adsl-63-207-238-99.dsl.chic01.pacbell.net) |
| 22:00.58 | *** part/#asterisk DaneM (n=DaneM@adsl-63-207-238-99.dsl.chic01.pacbell.net) |
| 22:03.02 | tdi | ok one more question, is it possible to make particular meetme conference (say in particular extension) use my own alicia sounds? |
| 22:03.14 | tdi | or should I replace alicia with sth else |
| 22:03.19 | tdi | globally |
| 22:04.32 | tdi | maybe by changing the lang .. :) |
| 22:05.17 | [TK]D-Fender | Assid: DETAILS |
| 22:05.56 | thekidrio | tdi, hrmm sounds like it should be doable. I am not that familiar with meetme though |
| 22:06.07 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 22:06.46 | lesouvage | JunK-Y: can you please give me one example on how to deal with this deprecated veriables or a link. |
| 22:07.26 | JacksLivr | when upgrading, do i need to specify any flags i set the first time? like the web-voicemail? |
| 22:09.39 | lesouvage | tdi: as far as I know you can add a directory in the var/lib/asterisk/sound/al directory and get the voiceprompt from their using Set(LANGUAGE()=al) before entering the conference. |
| 22:10.18 | lesouvage | tdi: I allready added the "al" subdir in my answer. |
| 22:11.36 | tdi | yep, i was thinking about such hack |
| 22:11.38 | tdi | thanks |
| 22:13.14 | Qwell[] | Defend: not I |
| 22:13.14 | Qwell[] | Defend: that'd be russellb |
| 22:13.20 | Qwell[] | send him money - lots of it |
| 22:13.32 | Qwell[] | russellb: I want 10% :P |
| 22:13.35 | Defend | o well him to :) i just saw your nick in the docs |
| 22:13.37 | file | I want 10.5% |
| 22:14.05 | Qwell[] | Defend: my name is in the sla docs? |
| 22:14.32 | Defend | its in the change logs talking about officaly supporting sla on ip trunks |
| 22:14.36 | Qwell[] | oh |
| 22:14.42 | Qwell[] | yeah, that |
| 22:15.32 | Defend | i dont know if it is worth anything but at one point we patched asterisk so when you park some one it sends an sms with the ext they were parked on so it was more like a key system |
| 22:15.42 | Defend | its a little dated now but |
| 22:15.51 | Defend | if ya all have any use for it |
| 22:17.56 | russellb | have you tried the new sla code yet? |
| 22:18.54 | JacksLivr | do this be the recommended method? http://users.pandora.be/Asterisk-PBX/UpdateAsterisk.htm |
| 22:19.11 | Qwell[] | no |
| 22:19.25 | Defend | nope not yet i use 1.4 on a little crap box but soon as my blades get here i will be putting 1.4 in production for the company so my guess would be like 2 weeks |
| 22:19.54 | *** join/#asterisk egypcio (n=egypcio@200.194.102.234) |
| 22:20.17 | Defend | i will play with it before hand of course but i prefer the load of the company to get good feedback |
| 22:20.44 | Defend | cause if something is wrong i will have 50 people yelling at me :) |
| 22:21.30 | Defend | thanks tho in advance for your work mate |
| 22:21.33 | creature1 | i am trying to run asterisk as an user called asterisk (non-root). I got it working with 1.4.0 but now after compiling and having installed 1.4.1 it doesnt seem to work as good. In Makefile i changed ASTVARRUNDIR to the following: 'ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk'. After starting asterisk i cannot connect to its remote console, says that asterisk.ctl isnt' found |
| 22:21.59 | creature1 | so, i launched it as root and see that asterisk.ctl gets created but as /var/run/asterisk.ctl |
| 22:22.22 | creature1 | isn't that strange? or am i missing something... |
| 22:22.29 | creature1 | most probably :D |
| 22:24.08 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 22:25.26 | creature1 | well... actually my question is how: where do i define where the asterisk.ctl and asterisk.pid is created? |
| 22:25.28 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
| 22:27.25 | mafkees | asterisk.conf |
| 22:27.56 | creature1 | mafkees: ah, thanks. |
| 22:27.58 | JacksLivr | should i use make upgrade or just install 1.4 over 1.2? I'm gonna start the compile here, so it'll be done when i get hom in a coupla hours. thanks. |
| 22:28.06 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 22:28.29 | mafkees | JacksLivr: I did: make uninstall in 1.2 and make install in 1.4 |
| 22:29.09 | creature1 | JacksLivr: you could do a make upgrade and it will download the newest source files. When all files are downloaded you can run make clean; make install. |
| 22:29.43 | JacksLivr | so either is as good. |
| 22:29.44 | creature1 | JacksLivr: and yes, if you had any special parameters you still want to use then you will have to specify them again as you recompile. |
| 22:29.49 | Qwell[] | no |
| 22:30.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:30.21 | Qwell[] | remove everything in /usr/lib/asterisk/modules/, get the 1.4 source, make, make install |
| 22:30.40 | file | Qwell[]: configure |
| 22:30.42 | JerJer | anyone ever notice that if a pure SIP call gets transferred around a lot the volume gets quieter and quieter after like 4 transfers ? |
| 22:30.47 | Qwell[] | umm, yeah, that |
| 22:30.50 | Qwell[] | but it'll tell you that :p |
| 22:30.55 | creature1 | :D |
| 22:31.00 | JacksLivr | k |
| 22:31.09 | JacksLivr | thanks |
| 22:31.35 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:31.51 | JacksLivr | gotta upgrade zaptel too |
| 22:32.15 | flenders | is there a way to transfer calls from the CLI? |
| 22:33.05 | Corydon-w | JerJer: what codecs is the call getting mashed through? |
| 22:33.11 | flenders | I mean, SIP/05 is on an outbound call, and wants to transfer the call to SIP/06. can that be done on the CLI? |
| 22:33.53 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 22:34.38 | mountainm2k | OK, I'm sure a bunch of people have asked this today, but how to adjust Polycom DST rules? |
| 22:34.47 | Qwell[] | see topic |
| 22:35.00 | Corydon-w | JerJer: I'd also be curious if the decibels were actually getting lower or if simply the tambre of the call was getting changed |
| 22:35.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:35.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:36.02 | JerJer | Corydon-w: There is a G.729 to ulaw conversion happening |
| 22:36.26 | JerJer | SIP Provider -> Phone is ulaw...then phone to phone would be G.729 |
| 22:37.28 | Corydon-w | True SIP transfer? Or multiple bridges? |
| 22:37.30 | JerJer | but it seems to happen with all ulaw as well |
| 22:37.54 | JerJer | tru sip transfers |
| 22:38.01 | Corydon-w | Odd |
| 22:38.16 | JerJer | snom ip phones |
| 22:38.18 | Corydon-w | I wouldn't have expected that |
| 22:38.42 | JerJer | running v1.2.14 |
| 22:39.10 | JerJer | i've tried to reproduce it here with 942's but can't |
| 22:40.04 | creature1 | JerJer: i have seen that aswell, using some old 1.2 version at that time |
| 22:41.00 | SplasPood | hrm... I know this is ghetto-rigged.. but I have a fax machine hooked up to a linksys PAP2 speaking SIP/RTP (g711 audio codec) over a local switched 100mbit link to an asterisk box, terminating those calls via a PRI attached directly to the asterisk box... Outbound faxing seems to work reasonably well.. Inbound faxing always fails and I get the "unknown RTP codec 100" warning... I know this isn't supposed to work *well* .. but I thought its possible |
| 22:41.00 | SplasPood | <PROTECTED> |
| 22:42.03 | Corydon-w | SplasPood: we had to set the device packet time to 10ms to get it to work reliably |
| 22:42.10 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 22:42.22 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:42.38 | Corydon-w | though we're using Azatel's, not PAP2's |
| 22:43.18 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:43.22 | JerJer | Corydon-w: the users state they feel the call is loosing decibles.... after a lot of transfering (7-8) they have to turn the volume up |
| 22:43.30 | JerJer | on the ip phone |
| 22:43.35 | FarrisG | I'm having some trouble with inbound DTMF accuracy on my sip trunks (from BandWidth.com). Any ideas? |
| 22:43.55 | JerJer | FarrisG: how do you get DTMF? |
| 22:43.59 | *** join/#asterisk zdrulio (n=krlozano@84.238.131.243) |
| 22:44.01 | SplasPood | Corydon-w: a setting on your azatel? |
| 22:44.10 | zdrulio | hello all |
| 22:44.26 | Corydon-w | SplasPood: yes |
| 22:44.40 | SplasPood | Corydon-w: hrm, we'll give that a whirl.. if I can find the same option |
| 22:44.43 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:44.52 | FarrisG | JerJer: The carrier tells us to use "inband", but it doesn't work at all unless I specify rfc2833. |
| 22:45.08 | JerJer | lovely |
| 22:45.17 | JerJer | don't use a compressed codec then |
| 22:45.43 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 22:46.40 | *** part/#asterisk toerkeium (i=oo@201.216.206.221) |
| 22:47.29 | *** part/#asterisk zdrulio (n=krlozano@84.238.131.243) |
| 22:49.44 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 22:50.23 | mercestes | FarrisG: Try dtmfmode=auto |
| 22:50.47 | SplasPood | Corydon-w: hrm, can't seem to locate an option that sounds like the one you mentioned |
| 22:51.04 | mercestes | FarrisG: Also, CanReinvite=yes and make sure you specify an Answer() before you plan to accept DTMF. |
| 22:52.15 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 22:56.00 | SplasPood | Corydon-w: what exactly does your option do.. maybe my naming is just a bit diff |
| 22:56.46 | ltdwk | SplasPood. I remember you |
| 22:57.08 | Corydon-w | Changes the packet size from 20ms (default) to 10ms |
| 22:57.23 | Corydon-w | So it sends double the packets, twice as fast |
| 22:57.37 | Corydon-w | (lower latency, really) |
| 22:57.48 | ltdwk | SplasPood: #linuxos, efnet. am I right? |
| 23:01.21 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 23:02.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:03.25 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 23:04.54 | Waverly360 | Guys..quick question. If I get a circuit/channel congestion error on a pri when I try to dial out, what's typically the problem? |
| 23:05.12 | Waverly360 | wanrouter doesn't give any errors at all when I start it up |
| 23:05.52 | JT | it usually means it's busy or similar :) |
| 23:05.57 | JT | fractional pri? |
| 23:06.19 | Waverly360 | yes |
| 23:06.28 | tzafrir_laptop | Is there any decent soft free phone phone for linux that supports video? |
| 23:06.43 | JT | Waverly360: you probably have the wrong channels set up |
| 23:06.51 | Waverly360 | JT: Hmm. |
| 23:06.54 | tzafrir_laptop | and can Asterisk work with Video in a partially decent way? |
| 23:06.59 | russellb | tzafrir_laptop: ekiga is the only one i know of |
| 23:07.45 | Waverly360 | JT: It's a fractional pri, channels 1-21, dchannel is 24 |
| 23:07.58 | Waverly360 | JT: we've confirmed with the provider that that's correct. |
| 23:08.02 | JT | Waverly360: what is the dial command? |
| 23:08.19 | Waverly360 | JT: Dial("SIP/5-08c87c58", "Zap/g2/5008043||tT") |
| 23:08.20 | JT | 21 out of 23 channels lit? weird |
| 23:08.26 | tzafrir_laptop | is it possible to have a sort of "menu" (like an IVR) of static images for video? |
| 23:08.52 | JT | Waverly360: try dialling G2 instead of g2 |
| 23:09.01 | JT | see if it makes a difference |
| 23:10.06 | russellb | tzafrir_laptop: asterisk doesn't support playback of mixed audio and video streams |
| 23:10.16 | russellb | tzafrir_laptop: but I guess you could just do playback/background of only video |
| 23:10.27 | russellb | that should work |
| 23:10.38 | tzafrir_laptop | how about a static image? |
| 23:10.58 | russellb | if the protocol supports sending an image file, then yeah |
| 23:11.18 | russellb | IAX2 does, but I don't know of a softphone that will accept images |
| 23:11.18 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 23:11.19 | russellb | gnophone used to ... but i don't know if that still works |
| 23:11.22 | russellb | that was years ago |
| 23:11.31 | russellb | and it doesn't support video |
| 23:11.45 | Waverly360 | JT: You really think that could be it? I've set it up on my other pbxes like that. |
| 23:12.12 | JT | Waverly360: it is a test, if you don't want to try it then feel free to do what you're already doing |
| 23:12.19 | JT | i have to head off for a bit |
| 23:12.32 | JT | if you want to do the test, then do it, if not, keep guessing :) |
| 23:12.55 | Waverly360 | heh |
| 23:12.56 | Waverly360 | I'll try it. |
| 23:13.00 | Waverly360 | Just doesn't make sense |
| 23:13.18 | JT | it does, i have solved the exact same problem before |
| 23:13.23 | JT | there could be a number of causes |
| 23:13.30 | JT | but you need to eliminate them one by one |
| 23:14.29 | Waverly360 | JT: That didn't fix it. |
| 23:16.21 | Waverly360 | Is there anyway to test whether the PRI is working without using asterisk? |
| 23:16.47 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:18.48 | Waverly360 | I gotta go for now, I may be back later. Thanks for the help JT |
| 23:20.36 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:20.59 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 23:21.05 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:21.48 | SplasPood | ltdwk: yea.. cept they all moved to slashnet.. efnet was years ago... who are you? |
| 23:22.08 | SplasPood | ltdwk: heh we're also in #rubyonrails together |
| 23:23.37 | *** join/#asterisk _lan_ (n=ww@201.2.77.83) |
| 23:24.30 | flenders | is there a way to transfer calls using the CLI or the manager api? |
| 23:24.42 | ltdwk | SplasPood: wildfire, wildy |
| 23:24.45 | ManxPower | flenders: not the CLI. Manager Redirect |
| 23:24.54 | ManxPower | see docs/manager.txt |
| 23:25.02 | flenders | ManxPower: thanks |
| 23:25.14 | tzafrir_laptop | hmmm... any idea how to change the sip port in ekiga? 5060 is used by Asterisk |
| 23:25.46 | SplasPood | ltdwk: ahh, I remember the nick |
| 23:25.57 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-193.cpe.netcabo.pt) |
| 23:26.03 | SplasPood | ltdwk: well they're all on irc.slashnet.org/#LinuxOS now |
| 23:26.16 | SplasPood | ltdwk: if you wanted to stop by... efnet #LinuxOS exists, but no one really there |
| 23:26.20 | ManxPower | tzafrir_laptop: ekiga is a linux SIP thingy? |
| 23:26.27 | ltdwk | SplasPood: those were good times mate |
| 23:26.47 | tzafrir_laptop | ManxPower, previesly knows as GnomeMeeting |
| 23:26.48 | ltdwk | SplasPood: that channel had real character |
| 23:27.07 | creature1 | tzafrir_laptop: http://wiki.ekiga.org/index.php/Internet_ports_used_by_Ekiga |
| 23:27.13 | ltdwk | SplasPood: were you around that time me and aphelion took over the channel? :) |
| 23:27.16 | lesouvage | exten => 1,1,Set(TIJDSTEMPEL=${STRFTIME()}) result in "TIJDSTEMPEL=Wed Mar 14 00:22:46 2007") Is there any format parameter so I get a result without spaces. |
| 23:28.08 | ManxPower | lesouvage: yes. see "man strftime" in a shell prompt or README.variables in the asterisk docs directory |
| 23:28.27 | tzafrir_laptop | If for this I need a gconfeditor, I'll try looking elsewhere |
| 23:28.32 | SplasPood | ltdwk: heh that must've been years ago.. but yea.. been there since day 1 in 1996 |
| 23:28.35 | creature1 | lesouvage: or, read more about it here http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 23:28.45 | lesouvage | thanks |
| 23:29.21 | ManxPower | Um, "show function strftime" would be the authortative source for info |
| 23:29.25 | creature1 | tzafrir_laptop: yepp you do |
| 23:29.33 | ManxPower | the Wiki is wrong our out of data much of the time |
| 23:29.59 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 23:30.35 | creature1 | tzafrir_laptop: if you do not want to fiddle with that, one client prefered by many users are http://gizmoproject.com/ |
| 23:31.02 | creature1 | tzafrir_laptop: easy to change ports in that client |
| 23:31.30 | tzafrir_laptop | creature1, non-free, and not for linux |
| 23:31.32 | *** join/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
| 23:31.53 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 23:32.03 | tzafrir_laptop | with e.g., twinkle, I didn't need to bother with setting the port number manually |
| 23:32.05 | ltdwk | SplasPood: What brings you to this neck of the woods.... dabbling in the ancient arts of telephony? |
| 23:32.06 | creature1 | tzafrir_laptop: isn't that one compatible with linux? |
| 23:32.31 | tzafrir_laptop | IIRC not. And anyway, it is non-fre |
| 23:32.38 | creature1 | tzafrir_laptop: it sure is |
| 23:32.47 | creature1 | (linux) |
| 23:32.47 | SplasPood | ltdwk: I've been idling here for over a year I guess.. I manage some asterisk installations |
| 23:32.47 | tzafrir_laptop | ~free |
| 23:32.48 | jbot | from memory, free is stuff might take awhile to get done |
| 23:33.10 | SplasPood | ltdwk: however right now I'm about to leave.. so.. catch me in #LinuxOS if you'd like.. otherwise, see ya around |
| 23:33.15 | stuntdouble | Have anyone successfully tried LDAP modules with Asterisk? I'm exploring and having issues with the oxymium and astirectory ones. |
| 23:33.16 | tzafrir_laptop | creature1, we have different definitions to the word "free" here |
| 23:33.25 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 23:33.41 | creature1 | http://www.gizmoproject.com/download.php |
| 23:33.45 | creature1 | Download Gizmo Project Now: It's Free! |
| 23:33.47 | creature1 | ;) |
| 23:34.04 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 23:35.06 | ltdwk | SplasPood: righto, ciao |
| 23:35.17 | creature1 | tzafrir_laptop: free for me is when you do not need to pay for using the software. |
| 23:35.22 | *** join/#asterisk interworx (n=weechat@203.220.28.66) |
| 23:36.18 | flenders | reload |
| 23:37.45 | rhombus | I have a system with a TDM-400 card in it and Polycom Soundpoint IP phones, and users are complaining about echo :( |
| 23:38.42 | *** join/#asterisk fatman2 (n=none@c-75-69-6-21.hsd1.vt.comcast.net) |
| 23:38.43 | fatman2 | <PROTECTED> |
| 23:38.53 | rhombus | I have used fxotune to tune the modules, and I have echo ratios of 0.7 and 3 percent, respectively |
| 23:39.48 | rhombus | I must admit I can hear it also, and it doesn't matter which channel the calls go out on |
| 23:40.12 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 23:40.27 | ltdwk | are you using zaptel echo cancellation? |
| 23:40.44 | ltdwk | 3% is quite easily cancelled |
| 23:40.44 | ManxPower | rhombus: what EC are you using? |
| 23:40.58 | *** part/#asterisk fatman2 (n=none@c-75-69-6-21.hsd1.vt.comcast.net) |
| 23:41.29 | *** join/#asterisk Opperior (n=chatzill@c-75-69-241-84.hsd1.nh.comcast.net) |
| 23:41.36 | Mahmoud | is it possible to put passwords whenever a user answers a phone call? |
| 23:42.04 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 23:42.09 | creature1 | Mahmoud: password for? |
| 23:42.10 | Mahmoud | i can easily place password prompts whenever a user wants to call outside numbers. But can I place password prompts for just answering phones? |
| 23:42.42 | Mahmoud | creativx, e.g. when the phone is ringing and you pic then phone. you are then prompted by a password, then connected to the person and can talk |
| 23:42.50 | Mahmoud | s/pic/pick/ |
| 23:43.28 | Mahmoud | problem is that i don't want any one to pick the phone, only people who know the password |
| 23:44.57 | Mahmoud | it can be done, but is it done so far? or just a features in the "wishlist" ? |
| 23:45.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:46.28 | ManxPower | Mahmoud: you would have to write complex dialplan logic involving Meetme and .call files |
| 23:46.42 | ManxPower | I doubt the M() option to dial would work for you |
| 23:47.00 | stuntdouble | Is LDAP a possible solution to use in a production environment to consolidate user/pass? Or should I rely on mysql or odbc? |
| 23:47.46 | *** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago) |
| 23:47.58 | flenders | I was having a look at: http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29 |
| 23:48.12 | flenders | and found out that I don't have the http.conf file |
| 23:48.28 | ManxPower | flenders: that would be a 1.4 and later feature |
| 23:48.28 | flenders | should I have done a make 'something' to install that? |
| 23:48.30 | Mahmoud | ManxPower, i wonder why app_meetme is not installed.. is this the it should be? |
| 23:48.35 | flenders | I'm running 1/4 |
| 23:48.37 | flenders | 1.4 |
| 23:48.44 | Mahmoud | ManxPower, lib/asterisk/modules doesn't have app_meetme.so |
| 23:48.45 | ManxPower | Mahmoud: meetme requires zaptel to be installed |
| 23:48.52 | Mahmoud | ManxPower, I see! |
| 23:48.56 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 23:49.05 | ManxPower | flenders: is there an httpd.conf.sample in the asterisk source configs directory? |
| 23:49.14 | flenders | nope |
| 23:49.19 | boch | is it possible to stop a Playback() with dtmf ? |
| 23:49.26 | Cyber-Dogg | so... I am curious about asterisk... |
| 23:49.34 | Cyber-Dogg | this is used for like automated telephone systems right? |
| 23:49.37 | ManxPower | boch: that is what Background is for |
| 23:49.42 | rhombus | ManxPower: I am using the default EC, whichever one that is |
| 23:49.51 | ManxPower | rhombus: try a different one |
| 23:50.13 | rhombus | ManxPower: I tried MG2, and it made it even worse |
| 23:50.21 | ManxPower | Cyber-Dogg: Asterisk is a PBX. One function of a PBX is IVR (menus) |
| 23:50.22 | rhombus | ManxPower: is there one you would suggest? |
| 23:50.31 | ManxPower | rhombus: one of the KB ones |
| 23:50.40 | Cyber-Dogg | what else does PBX allow? |
| 23:50.44 | Mahmoud | ManxPower, where can i find .call files? |
| 23:50.45 | rhombus | ManxPower: Have you ever been able to completely eliminate echo on analog lines? |
| 23:50.55 | ManxPower | Cyber-Dogg: making and receiving phone calls, voicemail. |
| 23:51.09 | ManxPower | rhombus: I don't use analog lines anymore. Too much trouble. |
| 23:51.11 | Cyber-Dogg | oh ok... |
| 23:51.25 | rhombus | ManxPower: What sort of trouble? |
| 23:51.27 | Cyber-Dogg | so this will work with VoIP and with the regular phone system too right? |
| 23:51.45 | ManxPower | rhombus: echo, dialing delays, callerid delays, locked modules |
| 23:51.53 | ManxPower | mostly locked TDM400P modules |
| 23:52.22 | ManxPower | Cyber-Dogg: generally asterisk would REPLACE the existing PBX (phone system) |
| 23:52.41 | rhombus | ManxPower: In other words, you've had echo problems too, and I take it you never resolved them |
| 23:52.41 | flenders | rhombus: I've been trying to get rid of echo on TDM400P for 4 weeks, and I couldn't |
| 23:52.55 | Cyber-Dogg | ManxPower: can I privmsg you? |
| 23:53.03 | ManxPower | rhombus: if you have CPU to spare and a small number of lines, you can try Digiums commercial EC module |
| 23:53.14 | flenders | rhombus: I still have echo during the first 2-3 seconds on a call, after that, it's gone |
| 23:53.21 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 23:53.22 | ManxPower | Cyber-Dogg: only if it is to let me know you have sent a paypay for private consulting time. |
| 23:53.33 | rhombus | flenders: convergence echo is not a problem -- the callers complain of constant echo, and I notice it too |
| 23:53.44 | rhombus | ManxPower: is that a separate card, or a module that plugs into the TDM-400? |
| 23:53.52 | ManxPower | rhombus: try echotraining=900 |
| 23:54.10 | ManxPower | rhombus: it is a commercial EC software module Digium licensed from a commercial EC company |
| 23:54.15 | Cyber-Dogg | I need to make a system that will call certain users as needed and ask them questions, record input and store it in a database |
| 23:54.21 | Cyber-Dogg | will Asterisk let me do that? |
| 23:54.32 | rhombus | ManxPower: we have only three lines |
| 23:54.39 | rhombus | ManxPower: maybe we should consider it |
| 23:54.42 | ManxPower | Cyber-Dogg: Asterisk will allow you to write/build an application to do that. |
| 23:54.48 | Cyber-Dogg | excellent |
| 23:54.50 | Cyber-Dogg | thanks! |
| 23:54.55 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 23:54.56 | ManxPower | rhombus: it would cost all of $30 for 3 lines |
| 23:55.09 | rhombus | ManxPower: cheap like borscht |
| 23:55.22 | ManxPower | rhombus: if the card is still under warrenty you might be able to get them for free. |
| 23:55.31 | rhombus | ManxPower: The card is new |
| 23:55.43 | flenders | ManxPower: that commercial EC software, is it this one: http://www.digium.com/en/products/software/hpec.php |
| 23:55.52 | ManxPower | we use Tellabs (external) EC for our T-1 PRIs |
| 23:55.55 | ManxPower | flenders: yes |
| 23:56.23 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:56.29 | rhombus | ManxPower: and have you been happy with the results? |
| 23:56.35 | *** part/#asterisk stuntdouble (n=ronald_l@082.143-60-66.DIA-subnet.surewest.net) |
| 23:58.12 | ManxPower | rhombus: Yes. Most of the Tellabs stuff is used and was originally installed at telcos (AT&T is what the sticker said) |
| 23:59.53 | ManxPower | New Tellabs EC is EXPENSIVE |